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

[Dafi Ni (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24146?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222676#comment-222676 ] 

Dafi Ni commented on ASTERISK-24146:
------------------------------------

I'm using different javascript library "JSSIP" 

steps for reproduce:
* install one of 12.5.0 , 12-svn, 13.0.0-beta1 asterisk version.
* generate certificates
* use attached configs from file reproduce-confs.zip
* try http://tryit.jssip.net/ with:
** name:153
** SIP uri: 153 at ASTERISK_IP_OR_DOMAIN
** SIP password: alamkota
** WS URI: wss://ASTERISK_IP_OR_DOMAIN:8089/ws
** where ASTERISK_IP_OR_DOMAIN could be asterisk ip address, or some domain pointing to this address
** certificate should be for ASTERISK_IP_OR_DOMAIN 
* open chrome or chromium browser point to https://ASTERISK_IP_OR_DOMAIN:8089/ws and accept certificate when self-signed or import CA certificate used for sign ASTERISK_IP_OR_DOMAIN certificate
* make call beetween ws->ws  wss->wss  or wss->udp clients with answer time greater than 7second





> 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, 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*



--
This message was sent by Atlassian JIRA
(v6.2#6252)