Thanks for detailed reply and sharing the valuable information. You are right i should also post this on discuss-webrtc forum, will do after i get a fresh call trace, possibly tomorrow.

Regarding your questions, yes the call establishes successfully at signalling level and audio stream works perfectly fine. I see DTLS certificates generated, sent, received, accepted etc. in RTPEngine logs. Both SIP and WebRTC clients successfully negotiate ICE and choose RTPEngine for media relay. However, the SIP client looses video after 1-2 seconds of call establishment, the audio still works during the entire call. So, i am convinced that problem is entirely related to video as far as the call establishment is concerned.

The WebRTC client is based on JSSIP but i am not aware of any config or method that may give us such control over media in response to this SIP INFO method. Any experienced JSSIP users / developers out there who may help on this?

The SIP client is based on a commercial SIP SDK. Both WebRTC and SIP clients work perfectly fine as long as other end is same, i.e. SIP to SIP and WebRTC to WebRTC video calls work as expected under same network conditions. Each client has only one video codec enabled, which is VP8, without any trans-coding etc. involved in between.

Thank you.



On Sun, Feb 1, 2015 at 8:04 AM, Gonzalo Gasca Meza <gascagonzalo@gmail.com> wrote:
Hi Muhammad

Can you comment if initially endpoints are receiving what is necessary to start sending Media at signaling level. For example: Successful ICE and SRTP-SDES/DTLS negotiation.

I see two issues here:
a) Establish a successful call
b) Once call is established how to deal with packet loss. Check this paper: http://static.googleusercontent.com/media/research.google.com/sv//pubs/archive/41611.pdf

From your email: "Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request?"

WebRTC in the browser depends on a upper transport layer in your case a SIP stack. Example: sipml5, sip.js, etc. Hence you need to modify that part there; so your signaling stack should interact with the Browser Media Engine upon recieving SIP INFO. 

Questions:
1. I would suggest to start a conversation in discuss-webrtc in Google Groups.
       -Which SIP stack are you using on the WebRTC client side?
       -Can you upload logs from WebRTC client and SIP client. (WebRTC/SIP/SIP stack)
       -Topology and browser version
       -Codec: VP8/H.264. This will help us to understand how media is handled.

If you do a packet capture can you still see Browser sending Video to SIP Client after those initial 5-7 seconds. (Check Webrtc logs/packet capture)

Some details about WebRTC handling packet loss.

https://groups.google.com/forum/#!topic/discuss-webrtc/0ZbxO05a9Zk

HTH

Thanks
-G




On Thu, Jan 29, 2015 at 2:56 PM, Muhammad Shahzad <shaheryarkh@gmail.com> wrote:
Hi,

This may be a bit out of focus topic for this forum but i am posting it here anyway with hope that some guru would shed some light on it and point me to right direction.

The problem is that i want to establish video call between a webrtc and a sip client using kamailio (for signalling) and RTPEngine (for media relay). Both signalling and the audio stream seems to work perfectly fine The remote video on webrtc client side (i.e. video stream from sip client) takes about 20-30 seconds to establish but once it starts it works fine. However, the remote video on sip client side (i.e. video stream from webrtc client) starts almost immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then it goes blank after about 30 seconds.

After a long discussion with sip client developer, we now understand the fact that sip client sends a request for so called key-frame, which is ignored by webrtc client. This request is sent through both RTCP stream and SIP INFO message.

The SIP INFO message seems to be pointless as media is internally managed by chrome/firefox and these browsers don't give us such sophisticated access and control over media streams. Please let me know if this assumption is wrong.

For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet type" error message in RTPEngine logs (not sure if it drops this packet or relay it anyway).

Does anyone has any idea on how can we either,

1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request?

OR

2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream on webrtc client's behalf?

If there is any other solution to this, please feel free to share.


Thank you.



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