[Spce-user] WebRTC to Legacy SIP no 2 way audio/video

[Jon Bonilla (Manwe)]

Hi all

Testing a WebRTC - Legacy SIP call here. I get different results depending on
how and what I try:

Tested with 2 different spce 3.8.2 servers.
Tested with Blink as legacy SIP client
Tested with tryit.jssip.net as WebRTC client (using wss port 1443) on Several
Firefox and Chrome versions. 

WebRTC user configured with "rtpproxy as only ICE candidate" and "encrypted sdp
with DTLS SRTP and RTCP feedback"

Legacy client configured as "plain SDP" and "RTP/AVP plain SDP"


What I get is:

- If Firefox is in the call, the spce removes firefox's video from sdp. Looks
  like it doesn't like it.

- Calling from webrtc to sip, I see the video offer and accept it but I get one
  way audio (from webrtc2sip) and no video in any direction. Tested with h264
  enabled and not.

- Calling from sip 2 webrtc I get one or two way audio (always works from
  webrtc2sip) depending on the chrome version I use.


I've been doing so many tests that I don't have captures of all, just some. But
I can repeat the tests and get captures and logs from every call. Just want to
ask if people are having the same issues as me.



thanks,

Jon


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