[asterisk-bugs] [JIRA] (ASTERISK-22127) Bridges/chan_sip: Directmedia settings not respected for setting up native_rtp bridge technology

[Jon Westgate (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22127?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=210466#comment-210466 ] 

Jon Westgate commented on ASTERISK-22127:
-----------------------------------------

I'd like to know why this is not a bug?
If it's not a bug, what is the correct way to disable native rtp_bridging on a per sip account basis?
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 native rtp bridge is never going to work.
however I would like native rtp bridge to work for internal calls.

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.

I guess there may actually be 2 bugs here:
1) Why does directmedia=no have no effect.
2) Why does rtp_bridge think it can bridge IPV4 and IPV6 rtp streams natively 

                
> Bridges/chan_sip: Directmedia settings not respected for setting up native_rtp bridge technology
> ------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22127
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22127
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp
>    Affects Versions: SVN, 12
>            Reporter: Jonathan Rose
>
> With two SIP phones A and B both with directmedia=no, A calls B with no feature flags.
> The bridge technology is native_rtp and SIP/A sends RTP directly to SIP/B and viceversa.
> The same behavior is observed directmedia=yes

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