WebRtc2SIP: No video is been received/transmitted when made call between chrome and a SIP client

StackOverflow https://stackoverflow.com/questions/21220147

Question

I am a newbie to webrtc2sip. I have setup my webrtc2sip gateway and registered to sip2sip.info as my domain. The problem is when I make video calls from chrome to any SIP client(ekiga/jitsi) the call gets connected but I am unable to see videos on both the sides.

================================================================================== Case 1: Chrome calls SIP client
Result: No video shown on both transmit and receive side

================================================================================== On the chrome JS console it says that :

State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=179:1
==session event = m_stream_video_local_added SIPml-api.js?svn=179:1
==session event = m_stream_video_remote_added SIPml-api.js?svn=179:1
==session event = m_stream_audio_local_added SIPml-api.js?svn=179:1
==session event = m_stream_audio_remote_added SIPml-api.js?svn=179:1

I have attached the JS console logs(case1_web2SIPClient_JSLogs.txt), wireshark trace(case1_web2SIPClient_WStrace.pcap) , webrtc2sip gateway console logs(case1_web2SIPClient_gatewayLogs.txt), sipml5 expert settings (Expert_settings.png) and config.xml (config.xml) for this case. I did not change anything in the config.xml that was generated after i built the source as mentioned in the instructions of this page (http://linux.autostatic.com/installing-webrtc2sip-on-ubuntu-1204).

I gave a try making calls between chrome and a android SIP client (CSipSimple) and the problem remains the same.

================================================================================== case 2: SIP client calling chrome.
Result: as soon as I click answer button on chrome, the calls gets rejected.

================================================================================== The JS console logs states that:

State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699 SIPml-api.js?svn=179:1
SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 172.21.128.118:10060;rport=10060;branch=z9hG4bK-1441398960
From: <sip:tata@172.21.229.127>;tag=300647977
To: <sip:amshyam320@sip2sip.info>;tag=ZxQFfM7fIIP3rT1HINzb
Call-ID: fbdf5a11-ff9e-0072-fa8b-09525220cec6
CSeq: 1670757835 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

For this case I am attaching JS logs(case2_SIPClient2WebJSLogs.txt), wireshark dump(case2_jitsiToWeb_WStrace.pcap)

Configuration:

Chrome Version: checked on 30.0.1599.114 and even on Latest chrome version Webrtc2sip version: 2.6.0 sipml5 Version: svn=203 ubuntu version: 12.04 (checked on both desktop and server editions)

Am I missing something in my setup or configuration please guide and help in moving further.

Thanks,
Shyam

Was it helpful?

Solution

Case2:

You're using RTCWeb-capable browser(Chrome) and trying to call a SIP client which may not be implementing some mandatory features like ICE,SRTP. Chrome uses SRTP-SDES and Firefox uses SRTP-DTLS.

Enable RTCWeb Breaker in sipml5 expert settings and check.

The RTCWeb Breaker is used to enable audio and video transcoding when the endpoints do not support the same codecs or the remote server is not RTCWeb-compliant.

Case:1: Is audio working? and I can't see your logs.

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