No audio in WebRTC and Asterisk

0
votes

I have a strange issue with Asterisk (in this case 13.2 version) and WebRTC.

So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local network - no nat or firewall.

The problem: if call is answered immediately - everything works fine. But if there are some delay in answer (say, 10 seconds) - no audio in both directions.

In RTP debug I saw that if there is some delay - destination ip address is incorrect. After removing ice servers from client config - both addresses is correct, but still no audio.

Below is debug for call with audio:
rtp http://pastebin.com/EzfByCG5
sip http://pastebin.com/1Y08yF5s

and no audio call (answered after 10 seconds delay):
rtp http://pastebin.com/TNnFkz6M
sip http://pastebin.com/zG7pjcZD

Also in FreeSwitch everything works fine, no matter when call is answered.

3
asterisksipwebrtcvoiprtp
webrtc is experemental protocol and require full understanding and advanced debuggin technics. Use freeswitch if it work fro you. – arheops
sure it works. But I love Asterisk and just want get it to work) anyway I'll post here link to my blog post if I've find any solution – loadaverage
Asterisk with IP based telephony (webRTC, SIP softphones) has been hit or miss for me. Each account had to be hand-crafted to match the client. Do yourself a favor, stick with Freeswitch for that. – Mantriur
You're right. Asterisk has some complexity in config sip accounts. But on other hand Asterisk can do many cool things - lke dynamically dialplan (even read all configuration from DB), REST api, AGI and AMI. All this is good for developers. PS: my experience with FreeSwitch is too small, maybe it can do this too :) – loadaverage

3 Answers

1
votes

Starting with Asterisk 12 you need to have pjproject libraries installed, otherwise you most likely won't have audio in your WebRTC calls and no warning whatsoever!

0
votes

First question: have you ensured all firewall configs permit RTP streams from the chosen STUN / ICE server unconditionally?

My own experience is that audio issues with WebRTC are almost always related to STUN / ICE & Firewall.

Did you follow a tutorial for your set up? If so, which one?

-1
votes

Check the log...

Peer audio RTP is at port 192.168.88.187:50026 sip_route_dump: route/path hop: sip:[email protected];transport=ws [Feb 13 06:15:59] ERROR[1055][C-00000031]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Feb 13 06:15:59] WARNING[1055][C-00000031]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'

I would use Freeswitch, no doubt :-)