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

[Vytis Valentinavičius (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22911?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=215388#comment-215388 ] 

Vytis Valentinavičius commented on ASTERISK-22911:
--------------------------------------------------

{quote}And to be fair, I have no idea in what scenario r405234 helped.{quote}

Let me think aloud:

The patch (r405234) cleared out locally stored remote candidates on create check list failure. So if we imagine that we are talking about a session with several REINVITEs, and the first INVITE created a valid ice session which resolved a path for media to travel, any subsequent requests to append the candidate list are useless since they will not be evaluated.

Now IF the first invite did not contain valid candidate pair, or all checks returned failure THEN appending the candidate list has at least some logic.

But either way, if first call to create check list succeeds, ice session is marked as started and the topmost 'if' returns early.

No, I still see no valid scenario where the patch in question fixed anything.

Maybe the person who created r405234 could explain more?
                
> [patch]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: Resources/res_pjsip, 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
>      Target Release: 11.8.0, 12.1.0
>
>         Attachments: capture_asterisk_211_client_15.pcap.gz, issue_22911.full.log, issue_22911.full.log, issue_22911.full.pjsip.log, works_on_my_machine.patch
>
>
> 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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