[asterisk-bugs] [JIRA] (ASTERISK-24146) No audio on WebRtc caller side when answer waiting time is more than ~7sec

[Aleksei Kulakov (JIRA)]

Aleksei Kulakov created ASTERISK-24146:
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             Summary: No audio on WebRtc caller side when answer waiting time is more than ~7sec
                 Key: ASTERISK-24146
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24146
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/WebSocket, Resources/res_rtp_asterisk
    Affects Versions: 12.4.0, 11.11.0
         Environment: Ubuntu 14.04

Asterisk 12.4.0 

PJProject(https://github.com/asterisk/pjproject 06/JUN/14)
--enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp CFLAGS="-g -DNDEBUG" 

chromium 35.0.1916.153(rev274914) (launch options: --use-fake-ui-for-media-stream --disable-webrtc-encryption)

SIPml-api.js?svn=224
            Reporter: Aleksei Kulakov


1. WebRtc caller(354) dials callee(6001) of any type
2. Callee waits 10sec before answering the call.
3. No audio on WebRtc caller(354) side, although RTP is flowing in both directions and callee can hear audio from caller mic.

There is some difference in output of 'rpt set debug on' in *bad* case(+answer wait time > 7sec+):
{quote}
Sent RTP P2P packet to 192.168.0.86:43911 (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.139:23506 (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.86:43911 (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.139:23506 (type 08, len 000160)
{quote}
and *good* case(+answer wait time <7sec+):
{quote}
Sent RTP P2P packet to 192.168.0.86:59092 (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.86:59092 (via ICE) (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.86:59092 (type 08, len 000160)
Sent RTP P2P packet to 192.168.0.86:59092 (via ICE) (type 08, len 000160)
{quote}

Issue reproducible only with chan_sip. *Chan_pjsip IS NOT affected*



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