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

[Dade Brandon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24146?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225697#comment-225697 ] 

Dade Brandon commented on ASTERISK-24146:
-----------------------------------------

Retracting my last comment about looking at this bug in a couple weeks;  it's definitely an asterisk bug, however sip.js has an option in their .invite method that allows you to invite without SDP,  which causes ice candidates to generate only after 2xx, right before sending an ack.  The invite without SDP process virtually guarantees that the ice/stun process will take less than eight seconds before the connection is made, thereby avoiding the bug.  

I'll cross post to the sip.js forum referenced a couple comments up

> 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: 11.11.0, 12.4.0
>         Environment: Ubuntu 14.04
> Asterisk 12.4.0 compiled from tarball
> 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
>         Attachments: badAsterDebug.log, badCall_filtered.pcapng, badChromeConsole.log, badChromeDebug.log, badChromeWebRtc.log, debug.zip, reproduce-confs.zip, sip.conf
>
>
> 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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