[asterisk-bugs] [JIRA] (ASTERISK-24806) Absent audio between WebRTC clients in local network

[Vadim (JIRA)]

Vadim created ASTERISK-24806:
--------------------------------

             Summary: Absent audio between WebRTC clients in local network
                 Key: ASTERISK-24806
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24806
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
    Affects Versions: 13.2.0
         Environment: Asterisk 13.2.0 on Linux 3.16.0-4-amd64, Debian GNU/Linux
Clients on Windows 7 Pro, 64-bit, Chrome 40
            Reporter: Vadim


I have a strange issue with Asterisk (in this case 13.2.0 version) and WebRTC.

So, I have latest Asterisk 13.2, latest Chrome (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, 10-15 seconds) - no audio in both directions.

In RTP debug I saw that if there is some delay - destination ip address is incorrect (it was the ip of network gateway). After removing ice servers from client config  both addresses have become correct, but still no audio.

Below is debug for call with audio:
RTP > http://pastebin.com/92A7Rxp2
SIP > http://pastebin.com/jagNVfgd
RTP+SIP > http://pastebin.com/8y87dLM7

and no audio call (answered after 10 seconds delay):
RTP > http://pastebin.com/r7mHdmCA
SIP > http://pastebin.com/jMX1zQze
RTP+SIP > http://pastebin.com/XE2CKN0E

Config files:
sip.conf > http://pastebin.com/eg9tr1A6
rtp.conf > http://pastebin.com/pGQB8WLh
extensions.conf > http://pastebin.com/1CiXhSmv
http.conf > http://pastebin.com/KFa3gLny

When audio is absent in RTP debug can be seen than "sent RTP packet" doesn't have "via ICE" mark. But when call initiator is any SIP client (X-Lite, Ekiga, etc) - WebRTC works perfectly.

In Sofia-SIP (SIP library for FreeSwitch) everything works fine, no matter when call is answered.
May this problem is caused by lack rtcp-mux in Asterisk?
Thanks.



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