webrtc2sip + sipml5 + asterisk no audio issue

Question

I am having a trouble in having a sipml5 to call other sipml5 via webrtc2sip and asterisk.

I have installed configured asterisk(version 11.10.0) + webrtc2sip(latest) + sipml5(chrome version 30.0.1599.66) to call from one box to other over websocket.

I can make a SIP call through and answer from other side but seems like there is no audio/voice packets gets exchanged as is evident by rtp and sip debug log and tcpdump.

asterisk and users are on separate servers and they do establish sip call but it's just a audio packet that does not appear.

I've created 2 users(1060 and 1061) and when I make a call I get these asterisk response.

rtp set debug on
RTP Debugging Enabled
*CLI>   == Spawn extension (default, 1060, 1) exited non-zero on 'SIP/1061-00000000'
== Using SIP RTP CoS mark 5
-- Executing [1060@default:1] Dial("SIP/1061-00000002", "SIP/1060") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1060
-- SIP/1060-00000003 is ringing
-- SIP/1060-00000003 answered SIP/1061-00000002

Asterisk settings are..

sip.conf

[general]
port=5060
bindaddr=0.0.0.0
context=default
transport=ws,wss,udp
srvlookup=yes

users.conf

[1060]
type=peer
username=1060
host=dynamic
secret=1234
context=default
disallow=all
allow=ulaw
transport=udp,ws,wss
encryption=yes
avpf=yes
icesupport=yes
nat=yes,force_rport

[1061]
type=peer
username=1061
host=dynamic
secret=1234
context=default
encryption=yes
avpf=yes
icesupport=yes
nat=yes
disallow=all
allow=ulaw
transport=udp,ws,wss

extensions.conf

[general]
static=yes
writeprotect=no

[default]
exten=>1060,1,Dial(SIP/1060)
exten=>1061,1,Dial(SIP/1061)

rtp.conf

[general]
icesupport=yes
stunaddr=stun.l.google.com:19302
strictrtp=no
rtcpinterval=6000
rtpchecksums=no

I can hear dialling sound on one end and ringing on the other end but as soon as the call is connected, can't hear anything.

Could anyone kindly help please.. I am very very desperate.. Thanks in advance!

webrtc is hily experemental now and have be used by expert only. — arheops
But I really need it working. it's just tough being an engineer. Please help.. — haeminish
So you need hire expert. Or enable debug and read it/teach yourself sip&webrtc — arheops

Fabrizio Lombardozzi Fabrizio Lombardozzi · Accepted Answer · 2016-05-14T11:00:51

You set rtp debug on but there's no rtp flow output, so call setup has gone wrong, you should check inside the dump of sip packets the sdp data they tried to share.
But you also have a better option: latest versions of Asterisk work with WebSocket and WebRTC like a charm without any external layer or component, so configuration will be much easier.
In addition you could use apache (at least 2.4) as websocket tunnel reverse proxy to the asterisk server, so you can do all the things (but rtp) with a single connection and a single ssl certificate use (another common problem is that the same selfsigned certificate is required to be explicitly accepted by user even if for the same server but on different port...and browser don't ask users to do that but just silently refuse to connect to websocket)

webrtc2sip + sipml5 + asterisk no audio issue

2 Answers