WebRTC Call thru asterisk

Asterisk Asterisk Support
1

Hi,
I have been trying to configure asterisk to do webrtc call.
Here are the system details:
Asterisk Version : Asterisk 13.5.0.
OS version : Linux ubuntu 3.16.0-30-generic
Browser : Chrome Version 45.0.2454.101 (64-bit)
FireFox 38.0.1

Problem : I could make audio call work between SIPML5 and a sip phone. (Xlite, Eyebeam)
But when I make a video call, call gets established but no video on both sides. Only audio is heard.
User details in sip.conf. (Assuming everything other configuration is ok as audio call is working fine)

[6001]
host=dynamic
secret=1234
context=from-internal
type=friend
encryption=true
avpf=true
force_avp=true
icesupport=true
directrtpsetup=yes
dtlsenable=true
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
videosupport=yes
disallow=all
allow = ulaw,h264,vp8
transport=tcp,ws

[6003]
host=dynamic
secret=1234
context=from-internal
type=friend
videosupport=yes
disallow = all
allow = ulaw,h263,h264
encryption=false
transport=udp,tcp

Debugging information. (I took some traces using wireshark)

  1. Call is initiated by webrtc client.
    2.Sdp exchange between sip phone and webrtc happens correctly. SIP phone receives ulaw/H263,H264
    in INVITE and responds with ulaw/H263.
  2. Webrtc side sends invite with ulaw/VP8 and receives ulaw/VP8 in 200OK from asterisk.
  3. When I looked at the RTP packets on sip phone side, The payload is dynamic and when I set filter
    in wireshark to interpret the payload 100 as vp8, I see vp8 packets coming from asterisk.
    (This I see as a problem. Not sure why it is not sending H263(a static payload type 34) and sending
    VP8. Doesnt asterisk do Video transcoding? If no, shouldnt it reject the call instead of sending
    proper sdp to sip side?)
  4. I also tried this on Firefox. Assuming that if phone and browser both can do H264, it should work.
    But no luck there too. Unfortunately I did not take any trace there.
  5. Even video between 2 SIPML extensions does not happen. Though call gets established with audio.
  6. I have tried setting directmedia to yes/no.

Am I missing some configuration on Asterisk side for video?
Any kind of debugging help on this will be a great thing.
Thank you,
Amol…

2

WebRTC use Vp8 only for video, Asterisk doesn’t support it unless you patch it. Since asterisk can’t transcode then you will need a GW to convert vp8 to h263(or patch asterik for VP8).

It is well know issue the video support between webrtc clients and asterisk withoput patches and without gateways in the middle.

3

As navaismo said, Asterisk cant transcode VP8, the VP8 video codec is used by Chrome