| Summary: | ASTERISK-24806: Absent audio between WebRTC clients in local network | ||||
| Reporter: | Vadim (sloun) | Labels: | |||
| Date Opened: | 2015-02-17 20:49:26.000-0600 | Date Closed: | 2015-02-18 11:43:47.000-0600 | ||
| Priority: | Major | Regression? | No | ||
| Status: | Closed/Complete | Components: | |||
| Versions: | 13.2.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Asterisk 13.2.0 on Linux 3.16.0-4-amd64, Debian GNU/Linux Clients on Windows 7 Pro, 64-bit, Chrome 40 | Attachments: | ( 0) answered_immediately.pcap ( 1) answered_with_delay.pcap | ||
| Description: | 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. | ||||
| Comments: | By: Vadim (sloun) 2015-02-17 21:23:48.566-0600 Wireshark pcap dump for call with delay in answer and for call answered immediately. | ||||