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

[Eugene Voityuk (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24146?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226487#comment-226487 ] 

Eugene Voityuk commented on ASTERISK-24146:
-------------------------------------------

Hi Again, new update, now i understand what is going on, maybe it will help you: I added some logic to ast_rtp_on_ice_complete callback, to see the status. Fist of all, calling dtls_perform_handshake() regardless of callback status is not good i think. Second, when you don't pickup in those magic time, the callback is called with next error message: "All ICE checklists failed (PJNATH_EICEFAILED)", and call will have no audio(i believe errors from here should be sent to console). If you picks up before that callback is fired, then you will have call with audio. Additional note, that Browsers set ICE to checking state only after they will have remoteDescription. So ICE checking should start after the ACK from caller to 200 OK, from callee. That will mean, that caller already have remote description, and will be in ICE checking state. Right now ice_start is called for any initial SDP with ICE candidates. Can anyone comment on my thoughts?

> 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, bad_call_client_and_server.zip, badCall_filtered.pcapng, badChromeConsole.log, badChromeDebug.log, badChromeWebRtc.log, debug.zip, logs_for_calls.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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