jssip (WEBRTC) SIP Signalling is successful but no RTP Delivered

[Phanindra Valluri]

We are making an outbound call from Webrtc(JSSIP) to PSTN through OVERSIP. Observed SIP Signalling is fine but no RTP is flowing.

Below steps are the sequence to do
1.Make an outbound call from WebRTC(Jssip) to a PSTN end point Via Oversip Proxy.
2.Audio cant be heard on both parties.
3.SIP Signalling is fine

We are expecting we should be hearing audio on both sides.

SIP Signalling is fine but there is dead air is flowing.

We are using Chrome , Firefox Latest versions on browser side , Oversip -2.0.4 .



We have implemented WEBRTC using the flavour of JSSIP. All SIP Signaling flows through the SIP Proxy(Opensource : OVERSIP) to the 

PSTN Network(Third Party Vendor).


Looking at a TCPDump of the local machine it does look like UDP (RTP) packets are not flowing in both directions. 

Observations:

1. There is no DTLS Packets flow in wireshark log.

2. ICEConnectionState is always in "checking" state and not moved to further state (chrome://webrtc-internals) .

Could you please help us in this regard.

Find the attached documents. 

[Iñaki Baz Castillo]

2016-12-21 16:52 GMT+01:00 Phanindra Valluri <vallur...@gmail.com>:
> Looking at a TCPDump of the local machine it does look like UDP (RTP)
> packets are not flowing in both directions.
>
> Observations:
>
> 1. There is no DTLS Packets flow in wireshark log.
>
> 2. ICEConnectionState is always in "checking" state and not moved to further
> state (chrome://webrtc-internals) .

You should check whether your SIP PSTN provider dupports WebRTC. This
is not something related to JsSIP (which uses the browser native
WebRTC engine) or OverSIP (which does not handle RTP at all).



--
Iñaki Baz Castillo
<i...@aliax.net>

[ravi raja]

Thanks for the quick reply. 

Our SIP PSTN provider supports WebRTC. Earlier the calls were successful.

[Iñaki Baz Castillo]

2016-12-21 17:19 GMT+01:00 ravi raja <ravira...@gmail.com>:
> Thanks for the quick reply.
>
> Our SIP PSTN provider supports WebRTC. Earlier the calls were successful.

Nice, but if there is any WebRTC related problem that's not due to
JsSIP or OverSIP.

[ravi raja]

We implemented the Webrtc product using (JSSIP, Oversip) and installed in one server. From some locations in US are unable to listen the voice even signaling is fine. Attached are the files for your reference.

[Iñaki Baz Castillo]

2016-12-21 17:29 GMT+01:00 ravi raja <ravira...@gmail.com>:
> We implemented the Webrtc product using (JSSIP, Oversip) and installed in
> one server. From some locations in US are unable to listen the voice even
> signaling is fine. Attached are the files for your reference.

Sorry, I won't work for free. Please, check it with your SIP PSTN provider.