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

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24806?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp closed ASTERISK-24806.
----------------------------------

    Resolution: Duplicate

> 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
>         Attachments: answered_immediately.pcap, answered_with_delay.pcap
>
>
> 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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