Asterisk: WebRTC no audio

[PinkerAffe]

I’m having what is probably a simple configuration issue. Calls between two SIP clients (zoiper) are successful.

When I start a call between a WebRTC client (sipml5) and a SIP client (Zoiper) is the connection active, but there is no audio in both directions available.

So, I have latest Asterisk 16, latest Chrome (with Firefox & Chrome Beta the same problem), sipml5 and a local network - no nat or firewall.

Any ideas on what we may be doing wrong? Thank you in advance for any help!

/etc/asterisk/sip.conf

`
[general]
realm=192.168.11.31;
udpbindaddr=192.168.11.31;
transport=udp

[1060] ; This will be WebRTC client
type=friend
username=1060
host=dynamic
secret=password
encryption=yes
avpf=yes
icesupport=yes
context=default
directmedia=no
transport=udp,ws,wss;
force_avp=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/cert/asterisk.pem
dtlssetup=actpass
rtcp_mux=yes

[6002] ; This will be the legacy SIP client
type=friend
username=6002
host=dynamic
secret=password
context=default

[6001]
type=friend
username=6001
host=dynamic
secret=password
context=default
`

/etc/asterisk/extensions.conf

[default]
exten => 1060,1,Dial(SIP/1060) exten => 6002,1,Dial(SIP/6002) exten => 6001,1,Dial(SIP/6001)

[jcolp]

The chan_sip module is not really maintained for WebRTC, chan_pjsip receives updates frequently and is known to work with JsSIP in Chrome. sipml5 should work as well with it, but it’s not something I personally test.

Otherwise you would need to isolate things further - having just the WebRTC client call Asterisk for example and doing a playback. You can then look at the ICE negotiation to see if a path was selected for media, and then go from there.