[SR-Users] NO_DTLS_FINGERPRINT when calling from webrtc to webrtc client

[Paweł Sternal]

Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-)

My problem is how to distinguish a call to WS UAC and how to SIP UAC in
scenarios:

1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC

2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2

WS kamailio is a proxy, SIP kamailio is a registrar

When "WS client1" is calling to "123123" WS kamailio doesn't know if
"123123" was registered from "WS client2" or from SIP UAC.

I have in this case rtpengine_manage("....... RTP/AVP"), but when INVITE
is returned to WS kamailio? RTP/SAVPF?

Probably it is obvious, however...

When WS client2 reply with 200OK, rtpengine_manage("..... ICE=force") to
WS client1 SDP is sent without a:fingerprint. sipml5 dumps warning:

message: "Could not negotiate answer SDP; cause = NO_DTLS_FINGERPRINT

I tried different combinations... and I'm stuck :/

Regards

[Daniel-Constantin Mierla]

Hello,

I would add the RTP-WebRTC gateway between SIP Kamailio and SIP UAC,
from resources point of view, it is the only leg that needs
encryption/decryption.

Otherwise, you can try to work with two rtpengine instances (sets) in WS
Kamailio, one to use for ws client to proxy and the other one for the
leg from proxy to ws client. It will be a communication between them
with classic rtp, both having towards ws client webrtc. It has the
drawback of decryption and encryption done two times for the same call.
You would need to add rtpengine set id in record-route to be able to
handle properly the re-INVITE, BYE, etc.

Another option that I would use is to send a negative reply from SIP
kamailio, catch that in failure_route in WS Kamailio and engace there
the rtpengine with proper flags. E.g., you assume it is going to be
webrtc-to-webrtc, so no encryption/decryption added first time invite
comes from WS client. You forward to SIP kamailio, which based on
location, if it discovers that the callee is classic SIP-RTP, will send
a 4xx back to WS Kamailio -- you end previous rtpengine session and
engage it again with new flags (use branch route for managing rtpengine
-- like it is done in default kamailio.cfg for rtpproxy).

Cheers,
Daniel

Post by Paweł Sternal
Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-)
My problem is how to distinguish a call to WS UAC and how to SIP UAC
1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC
2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2
WS kamailio is a proxy, SIP kamailio is a registrar
When "WS client1" is calling to "123123" WS kamailio doesn't know if
"123123" was registered from "WS client2" or from SIP UAC.
I have in this case rtpengine_manage("....... RTP/AVP"), but when
INVITE is returned to WS kamailio? RTP/SAVPF?
Probably it is obvious, however...
When WS client2 reply with 200OK, rtpengine_manage("..... ICE=force")
message: "Could not negotiate answer SDP; cause = NO_DTLS_FINGERPRINT
I tried different combinations... and I'm stuck :/
Regards
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[Paweł Sternal]

Thanks Daniel, you save my day :-) Two rtpengine solves my problem and
works perfect. This solution also adds me the possibility to record calls,
when between two rtpengine instances I will put rtpproxy.

Regards,
Pawel

Post by Daniel-Constantin Mierla
Hello,
I would add the RTP-WebRTC gateway between SIP Kamailio and SIP UAC, from
resources point of view, it is the only leg that needs
encryption/decryption.
Otherwise, you can try to work with two rtpengine instances (sets) in WS
Kamailio, one to use for ws client to proxy and the other one for the leg
from proxy to ws client. It will be a communication between them with
classic rtp, both having towards ws client webrtc. It has the drawback of
decryption and encryption done two times for the same call. You would need
to add rtpengine set id in record-route to be able to handle properly the
re-INVITE, BYE, etc.
Another option that I would use is to send a negative reply from SIP
kamailio, catch that in failure_route in WS Kamailio and engace there the
rtpengine with proper flags. E.g., you assume it is going to be
webrtc-to-webrtc, so no encryption/decryption added first time invite comes
from WS client. You forward to SIP kamailio, which based on location, if it
discovers that the callee is classic SIP-RTP, will send a 4xx back to WS
Kamailio -- you end previous rtpengine session and engage it again with new
flags (use branch route for managing rtpengine -- like it is done in
default kamailio.cfg for rtpproxy).
Cheers,
Daniel

Post by Paweł Sternal
Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-)
My problem is how to distinguish a call to WS UAC and how to SIP UAC in
1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC
2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2
WS kamailio is a proxy, SIP kamailio is a registrar
When "WS client1" is calling to "123123" WS kamailio doesn't know if
"123123" was registered from "WS client2" or from SIP UAC.
I have in this case rtpengine_manage("....... RTP/AVP"), but when INVITE
is returned to WS kamailio? RTP/SAVPF?
Probably it is obvious, however...
When WS client2 reply with 200OK, rtpengine_manage("..... ICE=force") to
message: "Could not negotiate answer SDP; cause = NO_DTLS_FINGERPRINT
I tried different combinations... and I'm stuck :/
Regards
_______________________________________________
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

[Daniel-Constantin Mierla]

Post by Paweł Sternal
Thanks Daniel, you save my day :-)

You're welcome! Great to see you ended up with more than expected :-)

Cheers,
Daniel

Post by Paweł Sternal
Two rtpengine solves my problem and works perfect. This solution also
adds me the possibility to record calls, when between two rtpengine
instances I will put rtpproxy.
Regards,
Pawel
Hello,
I would add the RTP-WebRTC gateway between SIP Kamailio and SIP
UAC, from resources point of view, it is the only leg that needs
encryption/decryption.
Otherwise, you can try to work with two rtpengine instances (sets)
in WS Kamailio, one to use for ws client to proxy and the other
one for the leg from proxy to ws client. It will be a
communication between them with classic rtp, both having towards
ws client webrtc. It has the drawback of decryption and encryption
done two times for the same call. You would need to add rtpengine
set id in record-route to be able to handle properly the
re-INVITE, BYE, etc.
Another option that I would use is to send a negative reply from
SIP kamailio, catch that in failure_route in WS Kamailio and
engace there the rtpengine with proper flags. E.g., you assume it
is going to be webrtc-to-webrtc, so no encryption/decryption added
first time invite comes from WS client. You forward to SIP
kamailio, which based on location, if it discovers that the callee
is classic SIP-RTP, will send a 4xx back to WS Kamailio -- you end
previous rtpengine session and engage it again with new flags (use
branch route for managing rtpengine -- like it is done in default
kamailio.cfg for rtpproxy).
Cheers,
Daniel
Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-)
My problem is how to distinguish a call to WS UAC and how to
1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC
2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2
WS kamailio is a proxy, SIP kamailio is a registrar
When "WS client1" is calling to "123123" WS kamailio doesn't
know if "123123" was registered from "WS client2" or from SIP UAC.
I have in this case rtpengine_manage("....... RTP/AVP"), but
when INVITE is returned to WS kamailio? RTP/SAVPF?
Probably it is obvious, however...
When WS client2 reply with 200OK, rtpengine_manage(".....
ICE=force") to WS client1 SDP is sent without a:fingerprint.
message: "Could not negotiate answer SDP; cause =
NO_DTLS_FINGERPRINT
I tried different combinations... and I'm stuck :/
Regards
_______________________________________________
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