[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=226494#comment-226494 ] 

Eugene Voityuk edited comment on ASTERISK-24146 at 6/15/15 8:17 AM:
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Good Day everyone here. I tried to implement what i told in previous comment, in asterisk code, and that made a trick, now ICE checking happens only after answer, and all WebRTC to SIP libraries works fine.. What i tested:
- SIP to SIP not affected
- SIP to WebRTC not affected
- WebRTC to SIP - now works as supposed, can wait until voicemail, forwarded, etc.
- WebRTC to WebRTC - works as supposed.

I am attaching patch files for this issue.
https://issues.asterisk.org/jira/secure/attachment/52531/res_rtp_asterisk.patch
https://issues.asterisk.org/jira/secure/attachment/52530/chan_sip.patch

I am waiting for license agreement sign confirmation. Then i believe we should review this on Gerrit? 


was (Author: sarumjanuch):
Good Day everyone here. I tried to implement what i told in previous comment, in asterisk code, and that made a trick, now ICE checking happens only after answer, and all WebRTC to SIP libraries works fine.. What i tested:
- SIP to SIP not affected
- SIP to WebRTC not affected
- WebRTC to SIP - now works as supposed, can wait until voicemail, forwarded, etc.
- WebRTC to WebRTC - works as supposed.

I am attaching patch files for this issue.
https://issues.asterisk.org/jira/secure/attachment/52531/res_rtp_asterisk.patch 

> 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, chan_sip.patch, debug.zip, logs_for_calls.zip, reproduce-confs.zip, res_rtp_asterisk.patch, 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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