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

Fri Sep 27 09:01:03 CDT 2013

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22564?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-22564:
-----------------------------------

    Description: 
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.

  was:
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:

– 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'

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:
[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

The phone's config
[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

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.

> 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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