[asterisk-bugs] [JIRA] (ASTERISK-22911) Asterisk fails to resume WebRTC call from hold

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22911?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212398#comment-212398 ] 

Matt Jordan commented on ASTERISK-22911:
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Just so we know how to try and reproduce this, you are running a version of Asterisk 12, using {{chan_sip}} with a websocket connection on one end of the bridge and a standard SIP over UDP channel on the other end of the bridge. The bridge chosen is a {{simple_bridge}}, as one channel is encrypting media using SDES-SRTP while the other is unencrypted.

>From a signalling perspective, there doesn't appear to be anything wrong here: Asterisk properly sends {{recvonly}} when the SIP over WS channel indicates {{sendonly}} and responds with {{sendrecv}} when the channel removes the Hold.

There are no unprotect errors in the log you supplied. I'm also not seeing any RTP errors or other indications of any issues in the log file.

You stated earlier that wireshark showed that Asterisk was sending RTP to the wrong port; it is clear that Chrome is changing the port number on the re-INVITE that initiates the hold. Can you attach a pcap demonstrating this?
                
> Asterisk fails to resume WebRTC call from hold
> ----------------------------------------------
>
>                 Key: ASTERISK-22911
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22911
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/WebSocket, Resources/res_rtp_asterisk
>    Affects Versions: 12.0.0-beta2
>         Environment: Server:
> asterisk:svn r403157  --with-srtp --with-pjproject
> pjproject:git asterisk/pjproject HEAD --with-external-srtp --enable-shared CFLAGS="-DNDEBUG"
> Ubuntu Precise 64, 3.2.0-23-generic.
> Client:
> Chrome 33.0.1720.0 canary
> http://sipml5.org/call.htm?svn=203
>            Reporter: Vytis Valentinavičius
>            Assignee: Vytis Valentinavičius
>         Attachments: issue_22911.full.log, issue_22911.full.log, issue_22911.full.pjsip.log
>
>
> When in call between soft-phone and WebRTC resuming from holden call does not resume the sound.
> Notices:
> 1. Hold and resume must be made by WebRTC client. Tested with sipml5.org demo.
> 2. Wireshark dump showed that after call is resumed all UDP packets do not reach WebRTC client due to wrong destination port.
> 3. Chrome stops active channel when issued hold command and creates new channel on resume. Channel is bound to new port each time.
> 4. Asterisk spits out such verbose errors:
> Before connection:
> [Nov 26 13:27:01] ERROR[2088]: pjsip:0 <?>: 	icess0x7fbe000 ..Error sending STUN request: Invalid argument
> Later in call (not related to Hold/Resume sequence):
> [Nov 26 13:28:06] WARNING[2177][C-00000000]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

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