Discussion:
[SR-Users] No audio/video transmission over different networks
Abhishek Saini
2014-09-03 10:23:57 UTC
Permalink
Hi,

I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video
calls seem to work well when both the devices are connected to the same
network, however, when one device connects to a different network (the two
devices now are on different networks), they are able to register on SIP
server, and even call can be triggered and accepted between the two devices
but there is no video/audio transmission.

I have setup rtpproxy but i don't know whether it's working or not.

Any help on this would be highly appreciated.


Following is my kamailio.cfg file:

#!define WITH_DEBUG
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_NAT
#!define WITH_USRLOCDB

tcp_async = yes
tcp_connect_timeout=5

#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.1 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
# block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL
DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default
'';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL
DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL
DEFAULT '';
#!endif

####### Include Local Config If Exists #########
import_file "kamailio-local.cfg"

####### Defined Values #########

# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://root:***@localhost/kamailio"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif

# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5

#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

####### Global Parameters #########

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif

memdbg=5
memlog=5

log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to disable the auto discovery of local aliases
based on reverse DNS on IPs (default on) */
auto_aliases=no

/* add local domain aliases */
alias="xyz.com"

/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available)
*/
#listen=udp:10.0.0.10:5060
listen=tcp:172.31.47.138:5060 advertise 54.191.193.***:5060

/* port to listen to
* - can be specified more than once if needed to listen on many ports */
port=5060

#!ifdef WITH_TLS
enable_tls=yes
#!endif

# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605

####### Custom Parameters #########

# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#

#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
pstn.gw_port = "" desc "PSTN GW Port"
#!endif

#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif

####### Modules Section ########

# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules/"
#!else
mpath="/usr/local/lib64/kamailio/modules/"
#!endif

#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif

loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"

#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif

#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif

#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif

#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif

#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif

#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif

#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif

#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif

#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif

#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif

# ----------------- setting module-specific parameters ---------------


# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")


# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)


# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)


# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)


# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif


# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif


# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)

# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif

#!endif


# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif


# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif


# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif


#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)

# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif


#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:xyz.com")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif


#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif

#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)

# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif

#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif

#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif

####### Routing Logic ########


# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {

# per request initial checks
route(REQINIT);

# NAT detection
route(NATDETECT);

# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}

# handle requests within SIP dialogs
route(WITHINDLG);

### only initial requests (no To tag)

t_check_trans();

# authentication
route(AUTH);

# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();

# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}

# dispatch requests to foreign domains
route(SIPOUT);

### requests for my local domains

# handle presence related requests
route(PRESENCE);

# handle registrations
route(REGISTRAR);

if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

# dispatch destinations to PSTN
route(PSTN);

# user location service
route(LOCATION);
}


route[RELAY] {

# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}

if (!t_relay()) {
sl_reply_error();
}
exit;
}

# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($sht(ipban=>$si)!=$null)
{
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
exit;
}
if (!pike_check_req())
{
xlog("L_ALERT","ALERT: pike blocking $rm from $fu
(IP:$si:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif

if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

if(!sanity_check("1511", "7"))
{
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and
discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}

# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();

exit;
}
}

# USER location service
route[LOCATION] {

#!ifdef WITH_SPEEDDIAL
# search for short dialing - 2-digit extension
if($rU=~"^[0-9][0-9]$")
if(sd_lookup("speed_dial"))
route(SIPOUT);
#!endif

#!ifdef WITH_ALIASDB
# search in DB-based aliases
if(alias_db_lookup("dbaliases"))
route(SIPOUT);
#!endif

$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}

# when routing via usrloc, log the missed calls also
if (is_method("INVITE"))
{
setflag(FLT_ACCMISSED);
}

route(RELAY);
exit;
}

# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;

if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {
route(TOVOICEMAIL);
# returns here if no voicemail server is configured
sl_send_reply("404", "No voicemail service");
exit;
}

#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
}

if(is_method("PUBLISH"))
{
handle_publish();
t_release();
} else if(is_method("SUBSCRIBE")) {
handle_subscribe();
t_release();
}
exit;
#!endif

# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}

# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH

#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif

if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}

#!endif
return;
}

# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop())
set_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}

# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;

rtpproxy_manage("co");

if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}

# URI update for dialog requests
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}

# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}

# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
return;
}

# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;

# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}

if (strempty($sel(cfg_get.pstn.gw_port))) {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
} else {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
+ $sel(cfg_get.pstn.gw_port);
}

route(RELAY);
exit;
#!endif

return;
}

# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif

# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE"))
return;

# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
return;
}
if(is_method("INVITE")) {
if($avp(oexten)==$null)
return;
$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
} else {
if($rU==$null)
return;
$ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
}
route(RELAY);
exit;
#!endif

return;
}

# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}

# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}

# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif

#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
$du = $null;
route(TOVOICEMAIL);
exit;
}
#!endif
}
Abhishek Saini
2014-09-04 11:01:37 UTC
Permalink
Hi Daniel,

Thanks, i was able to use the command you provided, but did not find the
chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by
calling from webrtc client to a desktop client(blink).

When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and has not much to do with the media(voice or video). So, that
means, it utilizes the rtpproxy to transmit the SIP signals(for
non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i
have always been able to make and receive calls and only the media (voice
or video) are not working (cross network).

I have also setup webrtc - it's working fine (firefox to firefox) but when
i call from firefox to desktop client, it does not work(only rings, but
does not connect).
I read about webrtc_breaker but there does not seem to be a module for that
in kamailio.

I think these two issues are somehow interlinked, please suggest me on
this.

Regards,
Abhishek
Hello,
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the
SDP. Also, the media IP in SDP should change from incoming INVITE to what
is sent out in the IP of rtpproxy.
Cheers,
Daniel
Regards,
Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <
Hello,
no time to look at config, but if you run the sip server on a private IP
behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
command line parameter.
Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and
video calls seem to work well when both the devices are connected to the
same network, however, when one device connects to a different network (the
two devices now are on different networks), they are able to register on
SIP server, and even call can be triggered and accepted between the two
devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Daniel-Constantin Mierla
2014-09-04 12:32:06 UTC
Permalink
Hello,

maybe you can send to mailing list the output of ngrep so we can look
and check if a rtp relay is used.

If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.

Cheers,
Daniel
Post by Abhishek Saini
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find
the chunks you have specified(a=nortproxy:yes (iirc)) in the data.
Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and has not much to do with the media(voice or video). So,
that means, it utilizes the rtpproxy to transmit the SIP signals(for
non-symmetric NAT), If so then i think, the rtpproxy is working fine,
as i have always been able to make and receive calls and only the
media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but
when i call from firefox to desktop client, it does not work(only
rings, but does not connect).
I read about webrtc_breaker but there does not seem to be a module for
that in kamailio.
I think these two issues are somehow interlinked, please suggest me on
this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla
Hello,
Post by Abhishek Saini
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you
have described (advertising address - found that after posting
the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in
the SDP. Also, the media IP in SDP should change from incoming
INVITE to what is sent out in the IP of rtpproxy.
Cheers,
Daniel
Post by Abhishek Saini
Regards,
Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla
Hello,
no time to look at config, but if you run the sip server on a
private IP behind a port forwarding address, you have to use
also rtpproxy with advertising address -- see the second
parameter of rtpproxy_manage() or search on the web for a
patch to rtpproxy to add advertising address via command line
parameter.
Cheers,
Daniel
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance.
The voice and video calls seem to work well when both the
devices are connected to the same network, however, when
one device connects to a different network (the two
devices now are on different networks), they are able to
register on SIP server, and even call can be triggered
and accepted between the two devices but there is no
video/audio transmission.
I have setup rtpproxy but i don't know whether it's
working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda
<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 -http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-04 13:24:50 UTC
Permalink
Hi,

Please find attached the output of ngrep for three type of
combinations/connections:

key: Blink is the desktop sip client and ntw means network.

blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful

We also need to enable webrtc to classic sip phone calls, like on
iphones/desktops etc. I could not find a good tutorial on rtpengine, and
the steps to replace rtpproxy with rtpengine.

Please suggest me on this.

Regards,
Abhishek
Hello,
maybe you can send to mailing list the output of ngrep so we can look and
check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.
Cheers,
Daniel
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the
chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by
calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and has not much to do with the media(voice or video). So, that
means, it utilizes the rtpproxy to transmit the SIP signals(for
non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i
have always been able to make and receive calls and only the media (voice
or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but
when i call from firefox to desktop client, it does not work(only rings,
but does not connect).
I read about webrtc_breaker but there does not seem to be a module for
that in kamailio.
I think these two issues are somehow interlinked, please suggest me on
this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla <
Hello,
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the
SDP. Also, the media IP in SDP should change from incoming INVITE to what
is sent out in the IP of rtpproxy.
Cheers,
Daniel
Regards,
Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <
Hello,
no time to look at config, but if you run the sip server on a private IP
behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
command line parameter.
Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and
video calls seem to work well when both the devices are connected to the
same network, however, when one device connects to a different network (the
two devices now are on different networks), they are able to register on
SIP server, and even call can be triggered and accepted between the two
devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-08 04:44:57 UTC
Permalink
Hi,

I have not heard on my last reply (it went in moderation). So, I am posting
one ngrep result here, Please let me know on this:

interface: any
filter: (ip or ip6) and ( port 5060 )
#
T 2014/09/04 12:51:26.423430 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
INVITE sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>.
Contact: <sip:48795260-***@public.gmane.org:1941;transport=tcp>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF.
a=sendrecv.

#
T 2014/09/04 12:51:26.424351 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias;received=182.64.39.131.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 INVITE.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:26.761371 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
ACK sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:27.144744 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
INVITE sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>.
Contact: <sip:48795260-***@public.gmane.org:1941;transport=tcp>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="admin", realm="abc.com",
nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc", uri="sip:hari-***@public.gmane.org",
response="f3286a5acf46bf90ae6962ab9fe37f25".
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF.
a=sendrecv.

##
T 2014/09/04 12:51:27.146564 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias;received=182.64.39.131.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:27.147226 172.31.47.138:5060 -> 182.64.39.131:59181 [A]
INVITE sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Record-Route: <sip:54.191.193.239;transport=tcp;lr=on;nat=yes>.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>.
Contact: <sip:48795260-***@public.gmane.org:1941
;transport=tcp;alias=182.64.39.131~3207~2>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7A
#
T 2014/09/04 12:51:27.147238 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
i8sovJF.
a=sendrecv.

#
T 2014/09/04 12:51:27.465870 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:27.475358 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 100 Trying.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:27.681136 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:27.839852 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 180 Ringing.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Contact: <sip:80523769-***@public.gmane.org:59180;transport=tcp>.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Content-Length: 0.
.

##
T 2014/09/04 12:51:27.840499 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 180 Ringing.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Contact: <sip:80523769-***@public.gmane.org:59180
;transport=tcp;alias=182.64.39.131~59181~2>.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Content-Length: 0.
.

#
T 2014/09/04 12:51:28.373579 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:30.264254 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
PUBLISH sip:admin-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f.
To: "abhishek" <sip:admin-***@public.gmane.org>.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 1 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/pidf+xml.
Content-Length: 761.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple
id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Aadmin%
40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple><dm:person
id="PID-8ee49af9e5034405300e43c6ecc590dd"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:30.264906 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias;received=182.64.39.131.
;tag=16061544dcc5db830f3ff5cfaeeb9db0.4dd8.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 1 PUBLISH.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:30.611704 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
PUBLISH sip:admin-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f.
To: "abhishek" <sip:admin-***@public.gmane.org>.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 2 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="admin", realm="abc.com",
nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX", uri="sip:admin-***@public.gmane.org",
response="ac284e634f819e5db0b8502e4ee8e8bf".
Content-Type: application/pidf+xml.
Content-Length: 761.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple
id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Aadmin%
40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple><dm:person
id="PID-8ee49af9e5034405300e43c6ecc590dd"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:30.612812 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 404 Not here.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias;received=182.64.39.131.
;tag=16061544dcc5db830f3ff5cfaeeb9db0.fc61.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 2 PUBLISH.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:31.146313 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:34.066168 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Contact: <sip:80523769-***@public.gmane.org:59180;transport=tcp>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.

##
T 2014/09/04 12:51:34.067096 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Contact: <sip:80523769-***@public.gmane.org:59180
;transport=tcp;alias=182.64.39.131~59181~2>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.

#
T 2014/09/04 12:51:34.408348 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp;alias=182.64.39.131~59181~2
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:34.409438 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:34.573751 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Contact: <sip:80523769-***@public.gmane.org:59180;transport=tcp>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.

##
T 2014/09/04 12:51:34.574344 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER.
Contact: <sip:80523769-***@public.gmane.org:59180
;transport=tcp;alias=182.64.39.131~59181~2>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.

#
T 2014/09/04 12:51:34.778979 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:34.912960 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp;alias=182.64.39.131~59181~2
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:34.913953 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:35.502496 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:36.094501 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
ACK sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

###
T 2014/09/04 12:51:44.484915 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
BYE sip:80523769-***@public.gmane.org:59180;transport=tcp;alias=182.64.39.131~59181~2
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Max-Forwards: 70.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:44.486679 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
BYE sip:80523769-***@public.gmane.org:59180;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Max-Forwards: 69.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:44.815901 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14957 BYE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

##
T 2014/09/04 12:51:44.816640 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
From: "abhishek" <sip:admin-***@public.gmane.org>;tag=93d6b660540f40338f9ed66f946ae11a.
To: <sip:hari-***@public.gmane.org>;tag=94d1930ec5b84d5bbd1241ce9feab364.
CSeq: 14957 BYE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:45.352809 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:47.241230 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
PUBLISH sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias.
Max-Forwards: 70.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=6ff49f144b63465ba09cf2152389be9d.
To: "Hari" <sip:hari-***@public.gmane.org>.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 1 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/pidf+xml.
Content-Length: 759.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple
id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Ahari%
40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple><dm:person
id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence>
##
T 2014/09/04 12:51:47.242107 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias;received=182.64.39.131.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=6ff49f144b63465ba09cf2152389be9d.
To: "Hari" <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.682e.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 1 PUBLISH.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:47.575172 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
PUBLISH sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias.
Max-Forwards: 70.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=6ff49f144b63465ba09cf2152389be9d.
To: "Hari" <sip:hari-***@public.gmane.org>.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 2 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="hari", realm="abc.com",
nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY", uri="sip:hari-***@public.gmane.org",
response="30700144d91c2d6f88b35b3dc6272170".
Content-Type: application/pidf+xml.
Content-Length: 759.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple
id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Ahari%
40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple><dm:person
id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:47.576515 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 404 Not here.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias;received=182.64.39.131.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=6ff49f144b63465ba09cf2152389be9d.
To: "Hari" <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.29b9.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 2 PUBLISH.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:47.954792 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:51.297577 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SUBSCRIBE sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias.
Max-Forwards: 70.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=bb028567e3c24d6792830754d4894f7d.
To: <sip:hari-***@public.gmane.org>.
Contact: <sip:80523769-***@public.gmane.org:59180;transport=tcp>.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14041 SUBSCRIBE.
Event: message-summary.
Expires: 600.
Supported: 100rel, replaces, norefersub, gruu.
Accept: application/simple-message-summary.
Allow-Events: conference, message-summary, dialog, presence,
presence.winfo, xcap-diff, dialog.winfo, refer.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.

#
T 2014/09/04 12:51:51.298522 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias;received=182.64.39.131.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=bb028567e3c24d6792830754d4894f7d.
To: <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.95ac.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14041 SUBSCRIBE.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:51.629211 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SUBSCRIBE sip:hari-***@public.gmane.org SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias.
Max-Forwards: 70.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=bb028567e3c24d6792830754d4894f7d.
To: <sip:hari-***@public.gmane.org>.
Contact: <sip:80523769-***@public.gmane.org:59180;transport=tcp>.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14042 SUBSCRIBE.
Event: message-summary.
Expires: 600.
Supported: 100rel, replaces, norefersub, gruu.
Accept: application/simple-message-summary.
Allow-Events: conference, message-summary, dialog, presence,
presence.winfo, xcap-diff, dialog.winfo, refer.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="hari", realm="abc.com",
nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N", uri="sip:hari-***@public.gmane.org",
response="a231fc305242b93cbdcbf58d36c67376".
Content-Length: 0.
.

#
T 2014/09/04 12:51:51.630383 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 404 No voicemail service.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias;received=182.64.39.131.
From: "Hari" <sip:hari-***@public.gmane.org>;tag=bb028567e3c24d6792830754d4894f7d.
To: <sip:hari-***@public.gmane.org>;tag=16061544dcc5db830f3ff5cfaeeb9db0.ddf9.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14042 SUBSCRIBE.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.

#
T 2014/09/04 12:51:52.001491 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......

Regards


On Thu, Sep 4, 2014 at 6:54 PM, Abhishek Saini <
Hi,
Please find attached the output of ngrep for three type of
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on
iphones/desktops etc. I could not find a good tutorial on rtpengine, and
the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla <
Hello,
maybe you can send to mailing list the output of ngrep so we can look and
check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.
Cheers,
Daniel
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the
chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by
calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and has not much to do with the media(voice or video). So, that
means, it utilizes the rtpproxy to transmit the SIP signals(for
non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i
have always been able to make and receive calls and only the media (voice
or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but
when i call from firefox to desktop client, it does not work(only rings,
but does not connect).
I read about webrtc_breaker but there does not seem to be a module for
that in kamailio.
I think these two issues are somehow interlinked, please suggest me on
this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla <
Hello,
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the
SDP. Also, the media IP in SDP should change from incoming INVITE to what
is sent out in the IP of rtpproxy.
Cheers,
Daniel
Regards,
Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <
Hello,
no time to look at config, but if you run the sip server on a private
IP behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
command line parameter.
Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and
video calls seem to work well when both the devices are connected to the
same network, however, when one device connects to a different network (the
two devices now are on different networks), they are able to register on
SIP server, and even call can be triggered and accepted between the two
devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-15 04:36:45 UTC
Permalink
Hi,

It appears that my last two messages have gone in moderation. Anyways, Can
you please tell me, how can i setup rtpengine on Ubuntu machine? After
installation - What configurations will i have to change?

I have lurked the internet a lot but did not find any tutorial on this.
Would appreciate any help on this.

Regards

On Mon, Sep 8, 2014 at 10:14 AM, Abhishek Saini <
Post by Abhishek Saini
Hi,
I have not heard on my last reply (it went in moderation). So, I am
interface: any
filter: (ip or ip6) and ( port 5060 )
#
T 2014/09/04 12:51:26.423430 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF.
a=sendrecv.
#
T 2014/09/04 12:51:26.424351 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias;received=182.64.39.131.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 INVITE.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:26.761371 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14955 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:27.144744 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="admin", realm="abc.com",
response="f3286a5acf46bf90ae6962ab9fe37f25".
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF.
a=sendrecv.
##
T 2014/09/04 12:51:27.146564 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias;received=182.64.39.131.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:27.147226 172.31.47.138:5060 -> 182.64.39.131:59181 [A]
Record-Route: <sip:54.191.193.239;transport=tcp;lr=on;nat=yes>.
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Max-Forwards: 69.
;transport=tcp;alias=182.64.39.131~3207~2>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Supported: 100rel, replaces, norefersub, gruu.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/sdp.
Content-Length: 639.
.
v=0.
o=- 3618843694 3618843694 IN IP4 192.168.0.3.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101.
c=IN IP4 192.168.0.3.
a=rtcp:50037.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:9 G722/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:103 speex/16000.
a=rtpmap:3 GSM/8000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT.
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7A
#
T 2014/09/04 12:51:27.147238 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
i8sovJF.
a=sendrecv.
#
T 2014/09/04 12:51:27.465870 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:27.475358 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 100 Trying.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:27.681136 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:27.839852 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 180 Ringing.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Content-Length: 0.
.
##
T 2014/09/04 12:51:27.840499 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 180 Ringing.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
;transport=tcp;alias=182.64.39.131~59181~2>.
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Content-Length: 0.
.
#
T 2014/09/04 12:51:28.373579 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:30.264254 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias.
Max-Forwards: 70.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 1 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/pidf+xml.
Content-Length: 761.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple
id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Aadmin%
40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple><dm:person
id="PID-8ee49af9e5034405300e43c6ecc590dd"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:30.264906 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias;received=182.64.39.131.
;tag=16061544dcc5db830f3ff5cfaeeb9db0.4dd8.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 1 PUBLISH.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:30.611704 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias.
Max-Forwards: 70.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 2 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="admin", realm="abc.com",
response="ac284e634f819e5db0b8502e4ee8e8bf".
Content-Type: application/pidf+xml.
Content-Length: 761.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple
id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Aadmin%
40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple><dm:person
id="PID-8ee49af9e5034405300e43c6ecc590dd"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:30.612812 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 404 Not here.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias;received=182.64.39.131.
;tag=16061544dcc5db830f3ff5cfaeeb9db0.fc61.
Call-ID: 100497fd86d14652a53347f3c5930ef3.
CSeq: 2 PUBLISH.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:31.146313 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:34.066168 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.
##
T 2014/09/04 12:51:34.067096 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
;transport=tcp;alias=182.64.39.131~59181~2>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.
#
T 2014/09/04 12:51:34.408348 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:34.409438 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:34.573751 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.
##
T 2014/09/04 12:51:34.574344 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias.
Record-Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 INVITE.
Server: Blink 0.9.1.2 (Windows).
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,
MESSAGE, REFER.
;transport=tcp;alias=182.64.39.131~59181~2>.
Supported: 100rel, replaces, norefersub, gruu.
Content-Type: application/sdp.
Content-Length: 352.
.
v=0.
o=- 3618843693 3618843694 IN IP4 192.168.0.6.
s=Blink 0.9.1.2 (Windows).
t=0 0.
m=audio 50000 RTP/AVP 113 101.
c=IN IP4 192.168.0.6.
a=rtcp:50001.
a=rtpmap:113 opus/48000.
a=fmtp:113 useinbandfec=1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1.
a=sendrecv.
#
T 2014/09/04 12:51:34.778979 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:34.912960 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:34.913953 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:35.502496 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:36.094501 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias.
Max-Forwards: 69.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14956 ACK.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
###
T 2014/09/04 12:51:44.484915 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Max-Forwards: 70.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
Route: <sip:54.191.193.239;transport=tcp;lr;nat=yes>.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:44.486679 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
Via: SIP/2.0/TCP 54.191.193.239:5060
;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;received=182.64.39.131;rport=3207;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Max-Forwards: 69.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:44.815901 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 54.191.193.239:5060
;received=54.191.193.239;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
##
T 2014/09/04 12:51:44.816640 172.31.47.138:5060 -> 182.64.39.131:3207 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.3:3207
;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias.
Call-ID: b0cac02326d44105aa008f9edd352e8e.
CSeq: 14957 BYE.
Server: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:45.352809 182.64.39.131:3207 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:47.241230 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias.
Max-Forwards: 70.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 1 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Type: application/pidf+xml.
Content-Length: 759.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple
id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Ahari%
40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple><dm:person
id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence>
##
T 2014/09/04 12:51:47.242107 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias;received=182.64.39.131.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 1 PUBLISH.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:47.575172 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias.
Max-Forwards: 70.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 2 PUBLISH.
Event: presence.
Expires: 600.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="hari", realm="abc.com",
response="30700144d91c2d6f88b35b3dc6272170".
Content-Type: application/pidf+xml.
Content-Length: 759.
.
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf"
xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple
id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3Ahari%
40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple><dm:person
id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence>
#
T 2014/09/04 12:51:47.576515 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 404 Not here.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias;received=182.64.39.131.
Call-ID: 47794de446374174a07d801f1f2a0965.
CSeq: 2 PUBLISH.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:47.954792 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
#
T 2014/09/04 12:51:51.297577 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias.
Max-Forwards: 70.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14041 SUBSCRIBE.
Event: message-summary.
Expires: 600.
Supported: 100rel, replaces, norefersub, gruu.
Accept: application/simple-message-summary.
Allow-Events: conference, message-summary, dialog, presence,
presence.winfo, xcap-diff, dialog.winfo, refer.
User-Agent: Blink 0.9.1.2 (Windows).
Content-Length: 0.
.
#
T 2014/09/04 12:51:51.298522 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias;received=182.64.39.131.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14041 SUBSCRIBE.
Proxy-Authenticate: Digest realm="abc.com",
nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N".
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:51.629211 182.64.39.131:59181 -> 172.31.47.138:5060 [AP]
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias.
Max-Forwards: 70.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14042 SUBSCRIBE.
Event: message-summary.
Expires: 600.
Supported: 100rel, replaces, norefersub, gruu.
Accept: application/simple-message-summary.
Allow-Events: conference, message-summary, dialog, presence,
presence.winfo, xcap-diff, dialog.winfo, refer.
User-Agent: Blink 0.9.1.2 (Windows).
Proxy-Authorization: Digest username="hari", realm="abc.com",
response="a231fc305242b93cbdcbf58d36c67376".
Content-Length: 0.
.
#
T 2014/09/04 12:51:51.630383 172.31.47.138:5060 -> 182.64.39.131:59181 [AP]
SIP/2.0 404 No voicemail service.
Via: SIP/2.0/TCP 192.168.0.6:59181
;rport=59181;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias;received=182.64.39.131.
Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389.
CSeq: 14042 SUBSCRIBE.
Server: kamailio (4.1.5 (x86_64/linux)).
Content-Length: 0.
.
#
T 2014/09/04 12:51:52.001491 182.64.39.131:59181 -> 172.31.47.138:5060 [A]
......
Regards
On Thu, Sep 4, 2014 at 6:54 PM, Abhishek Saini <
Hi,
Please find attached the output of ngrep for three type of
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on
iphones/desktops etc. I could not find a good tutorial on rtpengine, and
the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla <
Hello,
maybe you can send to mailing list the output of ngrep so we can look
and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.
Cheers,
Daniel
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find
the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked
by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and has not much to do with the media(voice or video). So, that
means, it utilizes the rtpproxy to transmit the SIP signals(for
non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i
have always been able to make and receive calls and only the media (voice
or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but
when i call from firefox to desktop client, it does not work(only rings,
but does not connect).
I read about webrtc_breaker but there does not seem to be a module for
that in kamailio.
I think these two issues are somehow interlinked, please suggest me on
this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla <
Hello,
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the
SDP. Also, the media IP in SDP should change from incoming INVITE to what
is sent out in the IP of rtpproxy.
Cheers,
Daniel
Regards,
Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <
Hello,
no time to look at config, but if you run the sip server on a private
IP behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
command line parameter.
Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and
video calls seem to work well when both the devices are connected to the
same network, however, when one device connects to a different network (the
two devices now are on different networks), they are able to register on
SIP server, and even call can be triggered and accepted between the two
devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-15 10:58:48 UTC
Permalink
Hi,

I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng
package on my ubuntu box. As suggested here:
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html

I have kept rtpproxy-ng's configuration same as the rtpproxy module, but
still not able to connect the webrtc calls to classic sip phones (and
vice-versa). Below is the sip message that is traced:


SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
From: "admin" <sip:admin-***@public.gmane.org>;tag=bzhwwG8nT2gFwwJgIyrz.
To: <sip:hari-***@public.gmane.org>;tag=OIllTQf.
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.

Can you please let me know, what's going wrong and how can i proceed.

Regards,
Abhishek
Daniel-Constantin Mierla
2014-09-15 12:49:15 UTC
Permalink
Hello,

the reply code indicates that the media type is not supported, thus
there has been no gatewaying between webrtc and classic rtp. Just
replacing rtpproxy with rtpengine is not enough, there are different
parameters that have to be provided.

Searching on web, I see that Carlos has published a config for it, see:
- https://github.com/caruizdiaz/kamailio-ws

Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have successfully setup rtpproxy-ng kamailio module and
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module,
but still not able to connect the webrtc calls to classic sip phones
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-16 07:45:10 UTC
Permalink
Hi Daniel,

Thanks for this.

I took the entire config files and configured it as per my ips and ports,
after doing that, still no call establishment(webrtc to classic sip phones
and vice-versa). Following is what i get in kamailio.log:

rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option
` '
ERROR: <script>: ==> duri=[sip:nudg.com:5060
;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option
` '
INFO: <script>: Reply from softphone: 100

And this SIP message:
SIP/2.0 603 Failed to get local SDP.

Regards,
Abhishek
Hello,
the reply code indicates that the media type is not supported, thus there
has been no gatewaying between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are different parameters that
have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but
still not able to connect the webrtc calls to classic sip phones (and
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-16 12:24:46 UTC
Permalink
Hi Daniel,


I was able to solve a fraction of my problem, Actually, the github link had
used rtpengine.so and i was using rptproxy-ng.so, there is a difference in
the flag conventions between the two; i modified that to achieve a little
progress.

Now, i am able to call on webrtc(firefox) from sip phone. However, after
accepting call, there is no audio, and disconnecting the call from either
end does not disconnect the call.

When i try to call from webrtc(firefox) to sip phone, there is no
signalling at all, and the sip phone to webrtc calls can't connect after
that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has
to be started again)

Following are the links to my latest kamailio.cfg file and port trace log
of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj

I am clueless at the moment!

Regards,
Abhishek



On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports,
after doing that, still no call establishment(webrtc to classic sip phones
rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option
` '
ERROR: <script>: ==> duri=[sip:nudg.com:5060
;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option
` '
INFO: <script>: Reply from softphone: 100
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <
Hello,
the reply code indicates that the media type is not supported, thus there
has been no gatewaying between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are different parameters that
have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module,
but still not able to connect the webrtc calls to classic sip phones (and
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Daniel-Constantin Mierla
2014-09-16 13:01:48 UTC
Permalink
Hello,

maybe you should play with kamailio master branch (which is in testing
phase before becoming 4.2) -- there you have the rtpengine -- and see
if you get it working. Once that, you can look at using an older
version, knowing you have it working and be able to compare. As I needed
latest features, whenever I needed webrtc gatewaying, I used devel
branch of rtpengine module.

Cheers,
Daniel
Post by Abhishek Saini
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github
link had used rtpengine.so and i was using rptproxy-ng.so, there is a
difference in the flag conventions between the two; i modified that to
achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However,
after accepting call, there is no audio, and disconnecting the call
from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no
signalling at all, and the sip phone to webrtc calls can't connect
after that. (I analyzed that mediaproxy-ng/rtpengine process
terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace
log of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards,
Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and
ports, after doing that, still no call establishment(webrtc to
classic sip phones and vice-versa). Following is what i get in
rtpp_test(): rtp proxy <udp:127.0.0.1:7722
<http://127.0.0.1:7722>> found, support for it enabled
unknown option ` '
ERROR: <script>: ==>
duri=[sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
unknown option ` '
INFO: <script>: Reply from softphone: 100
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla
Hello,
the reply code indicates that the media type is not supported,
thus there has been no gatewaying between webrtc and classic
rtp. Just replacing rtpproxy with rtpengine is not enough,
there are different parameters that have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have successfully setup rtpproxy-ng kamailio module and
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy
module, but still not able to connect the webrtc calls to
classic sip phones (and vice-versa). Below is the sip message
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 -http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-17 07:58:09 UTC
Permalink
Hi Daniel,

As you instructed, i installed kamailio from the master branch (which has
rtpengine module). Along with this, i installed the rtpengine package from
sipwise, as instructed by them.

I also updated this param : modparam("nathelper", "sipping_from", "
sip:pinger-***@public.gmane.org") to my domain

Now the scenario is as follows:

1) I am able to call webrtc(firefox and chrome) from iphone, the signalling
seems to be working fine, call can be paused, resumed etc.., but there is
no audio/video transmission.

2) Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.

Does it have any thing to do with rtp port ranges? or is there some other
misconfiguration?


Regards,
Abhishek
Hello,
maybe you should play with kamailio master branch (which is in testing
phase before becoming 4.2) -- there you have the rtpengine -- and see if
you get it working. Once that, you can look at using an older version,
knowing you have it working and be able to compare. As I needed latest
features, whenever I needed webrtc gatewaying, I used devel branch of
rtpengine module.
Cheers,
Daniel
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link
had used rtpengine.so and i was using rptproxy-ng.so, there is a difference
in the flag conventions between the two; i modified that to achieve a
little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after
accepting call, there is no audio, and disconnecting the call from either
end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no
signalling at all, and the sip phone to webrtc calls can't connect after
that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has
to be started again)
Following are the links to my latest kamailio.cfg file and port trace
log of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards,
Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and
ports, after doing that, still no call establishment(webrtc to classic sip
rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
ERROR: <script>: ==> duri=[
sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
INFO: <script>: Reply from softphone: 100
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <
Hello,
the reply code indicates that the media type is not supported, thus
there has been no gatewaying between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are different parameters that
have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module,
but still not able to connect the webrtc calls to classic sip phones (and
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-17 13:15:13 UTC
Permalink
Hi Daniel,

Here is something i traced in the log:

ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'
ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general
protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]

What's the cause of this error? i am using code from the master branch.
Perhaps this has something to do with the rptengine service
crash/termination.

Regards

On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has
rtpengine module). Along with this, i installed the rtpengine package from
sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", "
1) I am able to call webrtc(firefox and chrome) from iphone, the
signalling seems to be working fine, call can be paused, resumed etc.., but
there is no audio/video transmission.
2) Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other
misconfiguration?
Regards,
Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <
Hello,
maybe you should play with kamailio master branch (which is in testing
phase before becoming 4.2) -- there you have the rtpengine -- and see if
you get it working. Once that, you can look at using an older version,
knowing you have it working and be able to compare. As I needed latest
features, whenever I needed webrtc gatewaying, I used devel branch of
rtpengine module.
Cheers,
Daniel
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link
had used rtpengine.so and i was using rptproxy-ng.so, there is a difference
in the flag conventions between the two; i modified that to achieve a
little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after
accepting call, there is no audio, and disconnecting the call from either
end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no
signalling at all, and the sip phone to webrtc calls can't connect after
that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has
to be started again)
Following are the links to my latest kamailio.cfg file and port trace
log of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards,
Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and
ports, after doing that, still no call establishment(webrtc to classic sip
rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
ERROR: <script>: ==> duri=[
sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
INFO: <script>: Reply from softphone: 100
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <
Hello,
the reply code indicates that the media type is not supported, thus
there has been no gatewaying between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are different parameters that
have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Hi,
I have successfully setup rtpproxy-ng kamailio module and
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module,
but still not able to connect the webrtc calls to classic sip phones (and
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Abhishek Saini
2014-09-18 11:50:12 UTC
Permalink
Hi,

I have reported a bug on rptengine github, for the crash issue:
https://github.com/sipwise/rtpengine/issues/27

You mentioned that you have been using rtpengine kamailio module and the
rtpengine debian package with success. Was it on ubuntu box or some other
linux system? Sorry for asking this, but i am not able move ahead because
of this. Any other module suggestions from your end?

Thanks



On Wed, Sep 17, 2014 at 6:45 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'
ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general
protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]
What's the cause of this error? i am using code from the master branch.
Perhaps this has something to do with the rptengine service
crash/termination.
Regards
On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has
rtpengine module). Along with this, i installed the rtpengine package from
sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", "
1) I am able to call webrtc(firefox and chrome) from iphone, the
signalling seems to be working fine, call can be paused, resumed etc.., but
there is no audio/video transmission.
2) Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other
misconfiguration?
Regards,
Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <
Hello,
maybe you should play with kamailio master branch (which is in testing
phase before becoming 4.2) -- there you have the rtpengine -- and see if
you get it working. Once that, you can look at using an older version,
knowing you have it working and be able to compare. As I needed latest
features, whenever I needed webrtc gatewaying, I used devel branch of
rtpengine module.
Cheers,
Daniel
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link
had used rtpengine.so and i was using rptproxy-ng.so, there is a difference
in the flag conventions between the two; i modified that to achieve a
little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However,
after accepting call, there is no audio, and disconnecting the call from
either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no
signalling at all, and the sip phone to webrtc calls can't connect after
that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has
to be started again)
Following are the links to my latest kamailio.cfg file and port trace
log of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards,
Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <
Post by Abhishek Saini
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and
ports, after doing that, still no call establishment(webrtc to classic sip
rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
ERROR: <script>: ==> duri=[
sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
INFO: <script>: Reply from softphone: 100
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <
Hello,
the reply code indicates that the media type is not supported, thus
there has been no gatewaying between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are different parameters that
have to be provided.
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
Hi,
I have successfully setup rtpproxy-ng kamailio module and
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module,
but still not able to connect the webrtc calls to classic sip phones (and
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Daniel-Constantin Mierla
2014-09-04 07:04:48 UTC
Permalink
Hello,

no time to look at config, but if you run the sip server on a private IP
behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
command line parameter.

Cheers,
Daniel
Post by Abhishek Saini
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and
video calls seem to work well when both the devices are connected to
the same network, however, when one device connects to a different
network (the two devices now are on different networks), they are able
to register on SIP server, and even call can be triggered and accepted
between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
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