[asterisk-bugs] [JIRA] (ASTERISK-22564) Local bridge in bridge_native_rtp causes one way audio

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22564?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211058#comment-211058 ] 

Rusty Newton commented on ASTERISK-22564:
-----------------------------------------

@Jon,  Lots of work has happened on 12 since the 20th of Sep. Can you test with the latest SVN revision of 12 and report back, while also including the DEBUG log requested? (That is, be sure the log includes DEBUG messages enabled in logger.conf and turned up either via asterisk.conf or "core set debug 5")
                
> Local bridge in bridge_native_rtp causes one way audio
> ------------------------------------------------------
>
>                 Key: ASTERISK-22564
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22564
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp
>    Affects Versions: 12.0.0-alpha1
>         Environment: linux_x64 debian with Asterisk SVN-trunk-r399533
>            Reporter: Jon Westgate
>            Assignee: Jon Westgate
>         Attachments: debug.txt
>
>
> I note that this has been reported before and was closed as not a bug.
> If this is not a bug, what is the correct way to disable native rtp_bridging on a per sip account basis?
> It used to be canreinvite=no.
> I can observe on 2 systems running asterisk 12/SVN-trunk-r399533 that with directmedia=no set on both sip trunk and sip client that this happens:
> {noformat}
> – Executing [07974XXXXXX at wide:1] Dial("SIP/202-00000000", "SIP/JON_34XXXX/07974XXXXXX") in new stack
> == Using SIP RTP CoS mark 5
> – Called SIP/JON_34XXXX/07974XXXXXX
> > 0x7f0bdc018480 – Probation passed - setting RTP source address to [2XXX:XX0:0:X0::5060:2]:23569
> – SIP/JON_34XXXX-00000001 is ringing
> – SIP/JON_34XXXX-00000001 is making progress passing it to SIP/202-00000000
> > 0x7f0bdc018480 – Probation passed - setting RTP source address to [2XXX:XX0:0:X0::5060:2]:23569
> > 0x7f0bb003b470 – Probation passed - setting RTP source address to 8X.1X7.X85.X34:11794
> – SIP/JON_34XXXX-00000001 answered SIP/202-00000000
> – Channel SIP/202-00000000 joined 'simple_bridge' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
> – Channel SIP/JON_34XXXX-00000001 joined 'simple_bridge' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
> > Bridge 4ff3502f-2411-410e-9f9f-eda20b8b1efb: switching from simple_bridge technology to native_rtp
> – Channel SIP/JON_34XXXX-00000001 left 'native_rtp' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
> – Channel SIP/202-00000000 left 'native_rtp' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
> == Spawn extension (wide, 07974XXXXXX, 1) exited non-zero on 'SIP/202-00000000'
> {noformat}
> In this case I have a provider who I connect to via IPV6 and my ip-phone is IPV4-only so a remote rtp bridge is never going to work.
> My outgoing sip trunk:
> {noformat}
> [JON_34XXXX]
> type=peer
> remotesecret=password ; Our password to their service
> defaultuser=+44145XXXXXXX ; Authentication user for outbound
> host=voiceless.aa.net.uk
> canreinvite=no
> directmedia=no
> {noformat}
> The phone's config
> {noformat}
> [202]
> secret=xXxXxXxXxXxX
> username=202
> mailbox=202
> nat=force_rport,comedia
> type=friend
> context=wide
> host=dynamic
> canreinvite=no
> directmedia=no
> disallow=all
> allow=ulaw
> allow=gsm
> allow=alaw
> dtmfmode=rfc2833
> {noformat}
> Note that this not only effects asterisk in its IPV6 <> IPV4 bridging modes but also in IPV4 <> IPV4.
> I have a scenario where phones exist on a NAT network and the phone server is straddled between an external IP and the internal network.
> This causes all kinds of one way audio and dropped calls after random timings.

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