Discussion:
[SR-Users] SDPOPS issue or append_hf
Igor Potjevlesch
2014-08-06 09:56:34 UTC
Permalink
Hello,



I have an issue with the module SDPOPS while using
"sdp_keep_codecs_by_name".

If the calling party sends only one codec description like:



Content-Type: application/sdp

Content-Length: 202



v=0

o=UserA 2966746938 1790378070 IN IP4 10.141.0.21

s=Session SDP

c=IN IP4 10.141.0.21

t=0 0

m=audio 49152 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15



The result of the function "sdp_keep_codecs_by_name("PCMA,PCMU,G729a");" is:



Content-Type: application/sdp

Content-Length: 170

P-Asserted-Identity: "+0123456789" <sip:+0123456789-***@public.gmane.org>



v=0

o=UserA 2485672881 3000549892 IN IP4 a.b.c.d

s=Session SDP

c=IN IP4 a.b.c.d

t=0 0

m=audio 40330 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



If I open the capture in Wireshark, the PAI is not in the SDP part, and the
end of the capture after "a=rtpmap:8 PCMA/8000" is seen as "Data (18
bytes)".



I don't understand why the PAI is inserted within the SDP part. Adding the
PAI is done after "sdp_keep_codecs_by_name":



if (!is_present_hf("P-Asserted-Identity")) {

$var(pai) = $(fU{re.subst,/^0/+33/g});

append_hf("P-Asserted-Identity: \"$var(pai)\"
<sip:$var(pai)@$fd>\r\n");

}



I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.

Regards,



Igor.
Igor Potjevlesch
2014-08-06 10:23:24 UTC
Permalink
Hello,



To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.

But, the issue remains:



Content-Type: application/sdp

Content-Length: 169



v=0

o=UserA 1153072414 140968390 IN IP4 A.B.C.D

s=Session SDP

c=IN IP4 A.B.C.D

t=0 0

m=audio 60412 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



This SDP is dropped. Someone see something missing or wrong in the SDP
parts?

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 11:57
À : sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org
Objet : SDPOPS issue or append_hf



Hello,



I have an issue with the module SDPOPS while using
“sdp_keep_codecs_by_name”.

If the calling party sends only one codec description like:



Content-Type: application/sdp

Content-Length: 202



v=0

o=UserA 2966746938 1790378070 IN IP4 10.141.0.21

s=Session SDP

c=IN IP4 10.141.0.21

t=0 0

m=audio 49152 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15



The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is:



Content-Type: application/sdp

Content-Length: 170

P-Asserted-Identity: "+0123456789" <sip:+0123456789-***@public.gmane.org>



v=0

o=UserA 2485672881 3000549892 IN IP4 a.b.c.d

s=Session SDP

c=IN IP4 a.b.c.d

t=0 0

m=audio 40330 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



If I open the capture in Wireshark, the PAI is not in the SDP part, and the
end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18
bytes)”.



I don’t understand why the PAI is inserted within the SDP part. Adding the
PAI is done after “sdp_keep_codecs_by_name”:



if (!is_present_hf("P-Asserted-Identity")) {

$var(pai) = $(fU{re.subst,/^0/+33/g});

append_hf("P-Asserted-Identity: \"$var(pai)\"
<sip:$var(pai)@$fd>\r\n");

}



I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.

Regards,



Igor.
Daniel-Constantin Mierla
2014-08-06 14:42:12 UTC
Permalink
Hello,

the problem here is with rtpproxy marker -- can you try with the
parameter set to empty string?

- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856

Cheers,
Daniel
Post by Igor Potjevlesch
Hello,
To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.
Content-Type: application/sdp
Content-Length: 169
v=0
o=UserA 1153072414 140968390 IN IP4 A.B.C.D
s=Session SDP
c=IN IP4 A.B.C.D
t=0 0
m=audio 60412 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
This SDP is dropped. Someone see something missing or wrong in the
SDP parts?
Regards,
Igor.
*Envoyé :* mercredi 6 août 2014 11:57
*Objet :* SDPOPS issue or append_hf
Hello,
I have an issue with the module SDPOPS while
using “sdp_keep_codecs_by_name”.
Content-Type: application/sdp
Content-Length: 202
v=0
o=UserA 2966746938 1790378070 IN IP4 10.141.0.21
s=Session SDP
c=IN IP4 10.141.0.21
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
The result of the function
Content-Type: application/sdp
Content-Length: 170
v=0
o=UserA 2485672881 3000549892 IN IP4 a.b.c.d
s=Session SDP
c=IN IP4 a.b.c.d
t=0 0
m=audio 40330 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
If I open the capture in Wireshark, the PAI is not in the SDP part,
and the end of the capture after “a=rtpmap:8 PCMA/8000” is seen as
“Data (18 bytes)”.
I don’t understand why the PAI is inserted within the SDP part. Adding
if (!is_present_hf("P-Asserted-Identity")) {
$var(pai) = $(fU{re.subst,/^0/+33/g});
}
I guess that this cause my INVITE being dropped by 488 Media Not
Acceptable Here.
Regards,
Igor.
_______________________________________________
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://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
Igor Potjevlesch
2014-08-06 15:25:06 UTC
Permalink
Hello Daniel,



I got a feedback from the telco in the meantime. He told me that the issue
is the blank line between “rtpmap:8..” and “nortpproxy”.

This parameter is supported. I have successful calls with “nortpproxy=yes”.



I don’t know why sdp_keep_codecs_by_name inserts a blank line here.



Regards,



Igor.



De : sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org
[mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org] De la part de
Daniel-Constantin Mierla
Envoyé : mercredi 6 août 2014 16:42
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] SDPOPS issue or append_hf



Hello,

the problem here is with rtpproxy marker -- can you try with the parameter
set to empty string?

- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856

Cheers,
Daniel



On 06/08/14 12:23, Igor Potjevlesch wrote:

Hello,



To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.

But, the issue remains:



Content-Type: application/sdp

Content-Length: 169



v=0

o=UserA 1153072414 140968390 IN IP4 A.B.C.D

s=Session SDP

c=IN IP4 A.B.C.D

t=0 0

m=audio 60412 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



This SDP is dropped. Someone see something missing or wrong in the SDP
parts?

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 11:57
À : sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org <mailto:sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
Objet : SDPOPS issue or append_hf



Hello,



I have an issue with the module SDPOPS while using
“sdp_keep_codecs_by_name”.

If the calling party sends only one codec description like:



Content-Type: application/sdp

Content-Length: 202



v=0

o=UserA 2966746938 1790378070 IN IP4 10.141.0.21

s=Session SDP

c=IN IP4 10.141.0.21

t=0 0

m=audio 49152 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15



The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is:



Content-Type: application/sdp

Content-Length: 170

P-Asserted-Identity: "+0123456789" <sip:+0123456789-***@public.gmane.org>



v=0

o=UserA 2485672881 3000549892 IN IP4 a.b.c.d

s=Session SDP

c=IN IP4 a.b.c.d

t=0 0

m=audio 40330 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



If I open the capture in Wireshark, the PAI is not in the SDP part, and the
end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18
bytes)”.



I don’t understand why the PAI is inserted within the SDP part. Adding the
PAI is done after “sdp_keep_codecs_by_name”:



if (!is_present_hf("P-Asserted-Identity")) {

$var(pai) = $(fU{re.subst,/^0/+33/g});

append_hf("P-Asserted-Identity: \"$var(pai)\"
<sip:$var(pai)@$fd <sip:$var%28pai%29@$fd> >\r\n");

}



I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.

Regards,



Igor.






_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org <mailto:sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
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 ::: Oct 15-17, San Francisco, USA
Igor Potjevlesch
2014-08-06 15:48:42 UTC
Permalink
It’s really linked to the initial SDP. If I have only one codec, for example
G711u (plus telephone-event), and I just keep G711u, a blank line is
inserted.

If I keep G711u + telephone-event, everything is working fine.

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 17:25
À : miconda-***@public.gmane.org; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] SDPOPS issue or append_hf



Hello Daniel,



I got a feedback from the telco in the meantime. He told me that the issue
is the blank line between “rtpmap:8..” and “nortpproxy”.

This parameter is supported. I have successful calls with “nortpproxy=yes”.



I don’t know why sdp_keep_codecs_by_name inserts a blank line here.



Regards,



Igor.



De : sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org
<mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
[mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org] De la part de
Daniel-Constantin Mierla
Envoyé : mercredi 6 août 2014 16:42
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] SDPOPS issue or append_hf



Hello,

the problem here is with rtpproxy marker -- can you try with the parameter
set to empty string?

- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856

Cheers,
Daniel

On 06/08/14 12:23, Igor Potjevlesch wrote:

Hello,



To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.

But, the issue remains:



Content-Type: application/sdp

Content-Length: 169



v=0

o=UserA 1153072414 140968390 IN IP4 A.B.C.D

s=Session SDP

c=IN IP4 A.B.C.D

t=0 0

m=audio 60412 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



This SDP is dropped. Someone see something missing or wrong in the SDP
parts?

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 11:57
À : sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org <mailto:sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
Objet : SDPOPS issue or append_hf



Hello,



I have an issue with the module SDPOPS while using
“sdp_keep_codecs_by_name”.

If the calling party sends only one codec description like:



Content-Type: application/sdp

Content-Length: 202



v=0

o=UserA 2966746938 1790378070 IN IP4 10.141.0.21

s=Session SDP

c=IN IP4 10.141.0.21

t=0 0

m=audio 49152 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15



The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is:



Content-Type: application/sdp

Content-Length: 170

P-Asserted-Identity: "+0123456789" <sip:+0123456789-***@public.gmane.org>



v=0

o=UserA 2485672881 3000549892 IN IP4 a.b.c.d

s=Session SDP

c=IN IP4 a.b.c.d

t=0 0

m=audio 40330 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



If I open the capture in Wireshark, the PAI is not in the SDP part, and the
end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18
bytes)”.



I don’t understand why the PAI is inserted within the SDP part. Adding the
PAI is done after “sdp_keep_codecs_by_name”:



if (!is_present_hf("P-Asserted-Identity")) {

$var(pai) = $(fU{re.subst,/^0/+33/g});

append_hf("P-Asserted-Identity: \"$var(pai)\"
<sip:$var(pai)@$fd <sip:$var%28pai%29@$fd> >\r\n");

}



I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.

Regards,



Igor.





_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org <mailto:sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
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 ::: Oct 15-17, San Francisco, USA
Daniel-Constantin Mierla
2014-08-06 16:35:33 UTC
Permalink
It looks related to how changes are done to a sip message. rtpproxy is
working on incoming message as well as sdpops. Practically, rtpproxy
adds a new line at the end of the incoming sdp. sdopos deletes from old
sdp, resulting in empty lines inside the sdp.

Can you do the sdpops operation before record_route() and after it call
msg_apply_changes() from textopsx module?

Cheers,
Daniel
Post by Igor Potjevlesch
It’s really linked to the initial SDP. If I have only one codec, for
example G711u (plus telephone-event), and I just keep G711u, a blank
line is inserted.
If I keep G711u + telephone-event, everything is working fine.
Regards,
Igor.
*Envoyé :* mercredi 6 août 2014 17:25
*Objet :* RE: [SR-Users] SDPOPS issue or append_hf
Hello Daniel,
I got a feedback from the telco in the meantime. He told me that the
issue is the blank line between “rtpmap:8..” and “nortpproxy”.
This parameter is supported. I have successful calls with
“nortpproxy=yes”.
I don’t know why sdp_keep_codecs_by_name inserts a blank line here.
Regards,
Igor.
Daniel-Constantin Mierla
*Envoyé :* mercredi 6 août 2014 16:42
*À :* Kamailio (SER) - Users Mailing List
*Objet :* Re: [SR-Users] SDPOPS issue or append_hf
Hello,
the problem here is with rtpproxy marker -- can you try with the
parameter set to empty string?
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856
Cheers,
Daniel
Hello,
To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.
Content-Type: application/sdp
Content-Length: 169
v=0
o=UserA 1153072414 140968390 IN IP4 A.B.C.D
s=Session SDP
c=IN IP4 A.B.C.D
t=0 0
m=audio 60412 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
This SDP is dropped. Someone see something missing or wrong in
the SDP parts?
Regards,
Igor.
*Envoyé :* mercredi 6 août 2014 11:57
*Objet :* SDPOPS issue or append_hf
Hello,
I have an issue with the module SDPOPS while
using “sdp_keep_codecs_by_name”.
Content-Type: application/sdp
Content-Length: 202
v=0
o=UserA 2966746938 1790378070 IN IP4 10.141.0.21
s=Session SDP
c=IN IP4 10.141.0.21
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
The result of the function
Content-Type: application/sdp
Content-Length: 170
v=0
o=UserA 2485672881 3000549892 IN IP4 a.b.c.d
s=Session SDP
c=IN IP4 a.b.c.d
t=0 0
m=audio 40330 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
If I open the capture in Wireshark, the PAI is not in the SDP
part, and the end of the capture after “a=rtpmap:8 PCMA/8000” is
seen as “Data (18 bytes)”.
I don’t understand why the PAI is inserted within the SDP part.
if (!is_present_hf("P-Asserted-Identity")) {
$var(pai) = $(fU{re.subst,/^0/+33/g});
}
I guess that this cause my INVITE being dropped by 488 Media Not
Acceptable Here.
Regards,
Igor.
_______________________________________________
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 ::: Oct 15-17, San Francisco, USA
--
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 ::: Oct 15-17, San Francisco, USA
Igor Potjevlesch
2014-08-07 12:02:44 UTC
Permalink
I will try to look at this. It’s a bit tricky as the call-flow is not the
most easiest.

It’s strange because there is nothing to delete in that case because the
list of codecs is already okay.



Regards,



Igor.



De : Daniel-Constantin Mierla [mailto:miconda-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 18:36
À : Igor Potjevlesch; 'Kamailio (SER) - Users Mailing List'
Objet : Re: [SR-Users] SDPOPS issue or append_hf



It looks related to how changes are done to a sip message. rtpproxy is
working on incoming message as well as sdpops. Practically, rtpproxy adds a
new line at the end of the incoming sdp. sdopos deletes from old sdp,
resulting in empty lines inside the sdp.

Can you do the sdpops operation before record_route() and after it call
msg_apply_changes() from textopsx module?

Cheers,
Daniel

On 06/08/14 17:48, Igor Potjevlesch wrote:

It’s really linked to the initial SDP. If I have only one codec, for example
G711u (plus telephone-event), and I just keep G711u, a blank line is
inserted.

If I keep G711u + telephone-event, everything is working fine.

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 17:25
À : miconda-***@public.gmane.org <mailto:miconda-***@public.gmane.org> ; 'Kamailio (SER) - Users
Mailing List'
Objet : RE: [SR-Users] SDPOPS issue or append_hf



Hello Daniel,



I got a feedback from the telco in the meantime. He told me that the issue
is the blank line between “rtpmap:8..” and “nortpproxy”.

This parameter is supported. I have successful calls with “nortpproxy=yes”.



I don’t know why sdp_keep_codecs_by_name inserts a blank line here.



Regards,



Igor.



De : sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org
<mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
[mailto:sr-users-bounces-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org] De la part de
Daniel-Constantin Mierla
Envoyé : mercredi 6 août 2014 16:42
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] SDPOPS issue or append_hf



Hello,

the problem here is with rtpproxy marker -- can you try with the parameter
set to empty string?

- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856

Cheers,
Daniel

On 06/08/14 12:23, Igor Potjevlesch wrote:

Hello,



To be sure that the issue is not coming from append_hf, I add
(…,”Call-ID”). The PAI is now inserted after the Call-ID.

But, the issue remains:



Content-Type: application/sdp

Content-Length: 169



v=0

o=UserA 1153072414 140968390 IN IP4 A.B.C.D

s=Session SDP

c=IN IP4 A.B.C.D

t=0 0

m=audio 60412 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



This SDP is dropped. Someone see something missing or wrong in the SDP
parts?

Regards,



Igor.



De : Igor Potjevlesch [mailto:igor.potjevlesch-***@public.gmane.org]
Envoyé : mercredi 6 août 2014 11:57
À : sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org <mailto:sr-users-cR8azDVoa3IcDhw6gZKtMWD2FQJk+8+***@public.gmane.org>
Objet : SDPOPS issue or append_hf



Hello,



I have an issue with the module SDPOPS while using
“sdp_keep_codecs_by_name”.

If the calling party sends only one codec description like:



Content-Type: application/sdp

Content-Length: 202



v=0

o=UserA 2966746938 1790378070 IN IP4 10.141.0.21

s=Session SDP

c=IN IP4 10.141.0.21

t=0 0

m=audio 49152 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15



The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is:



Content-Type: application/sdp

Content-Length: 170

P-Asserted-Identity: "+0123456789" <sip:+0123456789-***@public.gmane.org>



v=0

o=UserA 2485672881 3000549892 IN IP4 a.b.c.d

s=Session SDP

c=IN IP4 a.b.c.d

t=0 0

m=audio 40330 RTP/AVP 8

a=rtpmap:8 PCMA/8000



a=nortpproxy:yes



If I open the capture in Wireshark, the PAI is not in the SDP part, and the
end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18
bytes)”.



I don’t understand why the PAI is inserted within the SDP part. Adding the
PAI is done after “sdp_keep_codecs_by_name”:



if (!is_present_hf("P-Asserted-Identity")) {

$var(pai) = $(fU{re.subst,/^0/+33/g});

append_hf("P-Asserted-Identity: \"$var(pai)\"
<sip:$var(pai)@$fd <sip:$var%28pai%29@$fd> >\r\n");

}



I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.

Regards,



Igor.






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Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
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