[asterisk-bugs] [JIRA] (ASTERISK-24253) Attended transfers with directmedia enabled sometimes set wrong rtp address

[Eli Hunter (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24253?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221914#comment-221914 ] 

Eli Hunter commented on ASTERISK-24253:
---------------------------------------

I've attached a full debug log calling 15555555555 (replaced the real phone number) from extension 313 then performing an attended transfer to 312 which gave dead air, and back to 313 which got audio working again.  

The debug log is rather busy since there's a lot of other endpoints registering and it's using a fairly complex ael script.  I replaced the asterisk server address with x.x.x.x and the endpoints are at 10.1.11.12 and 10.1.11.18.


> Attended transfers with directmedia enabled sometimes set wrong rtp address
> ---------------------------------------------------------------------------
>
>                 Key: ASTERISK-24253
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24253
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 12.4.0
>            Reporter: Eli Hunter
>            Assignee: Eli Hunter
>         Attachments: full_sample
>
>
> I've been testing directmedia with endpoints and see failures in about 1 out of 5 attended transfers.  It's using the endpoint's internal network address rather than the external IP.  The server is at a public IP and the endpoint is behind NAT.  I changed the IP the enpoint is behind to 1.1.1.1 and the sip provider's IP to 2.2.2.2.
> I thought it was the same as ASTERISK-23497 but it seems to be different and it wasn't fixed by upgrading from 12.2.0 to 12.4.0.
> Working transfer:
> {noformat}
>  == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/313
>     -- SIP/313-0000060a is ringing
>     -- SIP/313-0000060a answered SIP/312-00000609
>     -- Channel SIP/312-00000609 joined 'simple_bridge' basic-bridge <bfdd32fe-19a3-4438-9b46-59af26be886a>
>     -- Channel SIP/313-0000060a joined 'simple_bridge' basic-bridge <bfdd32fe-19a3-4438-9b46-59af26be886a>
>        > Bridge bfdd32fe-19a3-4438-9b46-59af26be886a: switching from simple_bridge technology to native_rtp
>        > 0x7fcde0015080 -- Probation passed - setting RTP source address to 1.1.1.1:2228
> Got  RTP packet from    1.1.1.1:2228 (type 09, seq 011738, ts 3706243606, len 000160)
> Sent RTP packet to      2.2.2.2:19500 (type 00, seq 012279, ts 2001056, len 000160)
> Sent RTP P2P packet to 1.1.1.1:2228 (type 00, len 000160)
> Got  RTP packet from    1.1.1.1:2228 (type 09, seq 011739, ts 3706243766, len 000160)
> Sent RTP packet to      2.2.2.2:19500 (type 00, seq 012280, ts 2001216, len 000160)
> {noformat}
> Failed transfer:
> {noformat}
>     -- Called SIP/312
>     -- SIP/312-00000608 is ringing
>     -- SIP/312-00000608 answered SIP/313-00000607
>     -- Channel SIP/313-00000607 joined 'simple_bridge' basic-bridge <36553219-70fa-46fa-a6cd-85b45e4e615b>
>     -- Channel SIP/312-00000608 joined 'simple_bridge' basic-bridge <36553219-70fa-46fa-a6cd-85b45e4e615b>
>        > Bridge 36553219-70fa-46fa-a6cd-85b45e4e615b: switching from simple_bridge technology to native_rtp
>        > 0x7fcde8705c30 -- Probation passed - setting RTP source address to 1.1.1.1:2226
>        > 0x7fcde8705c30 -- Probation passed - setting RTP source address to 1.1.1.1:2226
> Sent RTP packet to      10.1.11.18:2228 (type 09, seq 044259, ts 28978992, len 000160)
> Got  RTP packet from    2.2.2.2:19500 (type 00, seq 007364, ts 1178240, len 000160)
> Sent RTP packet to      10.1.11.18:2228 (type 09, seq 044260, ts 28979152, len 000160)
> {noformat}



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