[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=212303#comment-212303 ] 

Matt Jordan commented on ASTERISK-22911:
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Your log files don't illustrate the bug report.

In your first, issue_22911.full.log, two INVITE requests are received over the WS. In the first, the INVITE request is rejected due to the device not having registered. In the second, the INVITE request from the WS is accepted, but the target of the call returns a 486. This causes the call to be hung up.

Your second log file, issue_22911.full.pjsip.log, does not illustrate a call at all.

Please produce a DEBUG log illustrating the problem in the bug report. This should show the initial INVITE request over the WS and an established, bridged call. After the call is established, please show the hold taking place with restricted media via an INVITE request, followed by a subsequent INVITE request that attempts to take the call off hold.
                
> 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: 11.6.0, 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.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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