[SR-Users] Video Key-Frame Request using RTCP FIR or SIP INFO message

[Gonzalo Gasca Meza]

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 at 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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