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

[Iulius Ciorica (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24146?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225300#comment-225300 ] 

Iulius Ciorica commented on ASTERISK-24146:
-------------------------------------------

Hello all,

For *.pcap files:

https://groups.google.com/group/sip_js/attach/a4832fd73a490d64/no_audio.zip?part=0.1&authuser=0

10.10.14.140 is Asterisk 11.16 (web-rtc over wss)
10.10.10.179 is Google Chrome 40.0.2214.111 (SIP.js / sipML5 over wss)

Call Flow:

 Make call from Google Chrome(SIP.js - 1060) to any destination, it can be (SIP Softphone - 1061) or 3G mobile phone via ISDN E1 (TE820 Cards)

So, after some investigations, I found these:


 If I answer the call before ~8 seconds from initial invite, my call have audio in both ways:

 69 3.633843 10.10.14.140 10.10.10.179 DTLSv1.0 211 Client Hello
 
  User Datagram Protocol, Src Port: 17642 (17642), Dst Port: 53290 (53290)
 
 70 3.633845 10.10.14.140 10.10.10.179 DTLSv1.0 211 Client Hello
 
  User Datagram Protocol, Src Port: 17643 (17643), Dst Port: 53290 (53290)

Here you can see, source port are incremented and destination port are not incremented (53290 and 53290).


 If I answer the call after ~8 seconds from initial invite, my call do not have audio in any directions
 
 45 9.777806 10.10.14.140 10.10.10.179 DTLSv1.0 211 Client Hello

  User Datagram Protocol, Src Port: 16086 (16086), Dst Port: 55408 (55408)
 
 46 9.778873 10.10.14.140 10.10.10.179 DTLSv1.0 211 Client Hello
 
  User Datagram Protocol, Src Port: 16087 (16087), Dst Port: 55409 (55409)

Here you can see, source port are incremented and destination port are also incremented (55408 and 55409).
 
 47 9.778930 10.10.10.179 10.10.14.140 ICMP 239 Destination unreachable (Port unreachable)
 
  Type: 3 (Destination unreachable)
  Code: 3 (Port unreachable)
 
 and so on...

References:

https://groups.google.com/forum/#!topic/sip_js/Gl_kkGUfHCo

Thank you,
Iulius

> 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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