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Summary:ASTERISK-15661: One way audio after placing call on hold and resuming
Reporter:Jordan Kirby (jordankirby)Labels:
Date Opened:2010-02-19 09:21:25.000-0600Date Closed:2011-06-07 14:07:24
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) full.txt
( 1) OnHoldTest.pcap
( 2) sip_debug.txt
( 3) sip_show_channels.txt
( 4) sip_show_history.txt
Description:This fault occurs on 1.6.1.11, 1.6.2.0, 1.6.2.2, 1.6.2.3-RC2, SVN-247894.

sip.conf: directrtpsetup=yes, nat=yes
Both extensions: canreinvite=yes, nat=yes
Phones are on the same LAN and behind NAT.

Server is in a separate location also behind NAT. All standard internal and external calls work fine.

Problem:

Extension A calls extension B. Extension A puts the call on hold, extension B gets played music as expected. When extension A resumes the call extension B can't hear extension A.

This seems to be because Asterisk sends the external IP address of extension B to extension A in the SDP when the call is resumed.

****** ADDITIONAL INFORMATION ******

I will attach a wireshark trace and copies of the "sip show channel xxx" results for each leg of the call.


Comments:By: Jordan Kirby (jordankirby) 2010-02-19 09:22:57.000-0600

The same problem occurs when the server is not behind NAT (phones still NATd).
Attached traces are from a server not behind NAT.
By: Jordan Kirby (jordankirby) 2010-02-19 09:27:25.000-0600

The incorrect Audio IP seems to sometimes be the IP of the server and sometimes the external IP of phone A.
By: Leif Madsen (lmadsen) 2010-02-19 12:36:43.000-0600

Can you also provide a SIP trace from the Asterisk console, along with SIP history and debug level logging (as set from logger.conf)?
By: David Woolley (davidw) 2010-02-22 05:59:39.000-0600

I wonder if this is related to ASTERISK-1345545.  The fix for that was to prevent partial implementation of directrtpsetup when directrtpsetup wasn't enabled.  However, I believe it would still be a problem if directrtpsetup were enabled.
By: Jordan Kirby (jordankirby) 2010-02-22 07:58:46.000-0600

I have uploaded a copy of the full log (with debug on), the asterisk sip debug and the history from the channels I could see after the hold and resume.
By: Russell Bryant (russell) 2010-09-16 11:41:57

We were able to reproduce this in an older version but not in current versions, so I think this is fixed.  Can you please confirm?
By: Jordan Kirby (jordankirby) 2010-10-28 03:32:53

I have retested this issue on 1.6.2.13 and 1.8.0 and I can confirm the problem still exists.
By: Leif Madsen (lmadsen) 2010-11-04 14:03:04

Do you have externip set? It sounds like you might.

Do you have localnet set? If not, that is likely the problem.

Sounds like it might be a configuration issue in sip.conf.