[asterisk-bugs] [JIRA] (ASTERISK-24253) Attended transfers with directmedia enabled sometimes set wrong rtp address
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Thu Sep 25 16:29:29 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24253?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton closed ASTERISK-24253.
-----------------------------------
Resolution: Suspended
I'm closing this out, as there hasn't been a response for a while and we still need further information to determine if this is a bug or not.
It sounds like a bug, but I can't reproduce and I can't find any other reports of the issue. So, the developers don't have a way to reproduce on their systems for testing and diagnosing.
If someone wants to re-open this issue, then we need the following information:
* A guide to reproduction including
** sip.conf
** extensions.conf
** Any other configs required, and steps for reproduction.
* Logs/debug
** PCAP demonstrating the issue
** Asterisk log with VERBOSE and DEBUG correlating to the PCAP.
If you need to talk with a bug marshal or need help understanding how to get this information you can find us on irc.freenode.net in #asterisk-bugs and #asterisk-dev.
Thanks!
> 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: Matt Jordan
> 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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