samohan
samohan

Reputation: 109

Asterisk12 and sipML5 video support

is it possible to send video each other on asterisk12 and sipML5 demo site? I can hear audio but video is black screen... I set sip.conf like this,

[general]
videosupport=yes
[6001]
host=dynamic
secret=1234
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h261
allow=h263
allow=h264
allow=vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

[6002]
host=dynamic
secret=1234
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h261
allow=h263
allow=h264
allow=vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

my environment, sipml5 demo site http://sipml5.org/call.htm?svn=224

asterisk Asterisk 12.4.0 built by root @ 48asterisk on a x86_64 running Linux on 2014-08-01 08:22:18 UTC

OS debian 7.6

Thank you for your cooperation.

Upvotes: 2

Views: 3683

Answers (2)

hamedmehryar
hamedmehryar

Reputation: 414

1- in asterisk you should allow only ONE video codec for each peer e.g:

disallow=all

allow=h263

as asterisk does not support video codec negotiation! 2-you'd better use webrtc2sip (http://webrtc2sip.org/) between sipml5 and asterisk. it solves protocol negotiation issues such as SRTP and ICE are mandatory in webrtc specification. it also does codec conversion between multiple browsers.

i have set a sipml5-webrtc2sip-asterisk structure and it works very fine!! :)

Upvotes: 1

AlexGreg
AlexGreg

Reputation: 838

make sure that you have enabled websocket in http.conf and configure sipml to point to your WS here. Plus, your peers aren't configured with ws transport type.

If you follow this guide step by step it'll all work...

Upvotes: 1

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