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Summary:ASTERISK-23182: SIP/OK response to a SIP/BYE request is sent to the wrong port
Reporter:Marko Seidenglanz (markose)Labels:
Date Opened:2014-01-24 03:18:28.000-0600Date Closed:2014-02-10 08:05:00.000-0600
Priority:TrivialRegression?No
Status:Closed/CompleteComponents:Resources/res_pjsip
Versions:12.0.0 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When a SIP/BYE request is sent to Asterisk it sends it's SIP/OK to the NAT-Port it has received the message from. The problem is, that this message never arrives at the destination:


<--- Received SIP request (414 bytes) from UDP:23.251.129.236:51723 --->
BYE sip:3e315314-9519-450e-a0fe-45912f493819@213.71.23.139:5060 SIP/2.0
Via: SIP/2.0/UDP wh1.24dial.com:5060;rport;branch=z9hG4bK123456789
From: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com:5060>;tag=123456789
To:  <sip:018053557555@213.71.23.139>;tag=ae356e6b-e90e-457d-9c17-8689f75eafd9
Contact: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com:5060>
Call-ID: 4a75f291-3b82-4920-ad1e-2d43360ebeff
CSeq: 24500 BYE


<--- Transmitting SIP response (337 bytes) to UDP:23.251.129.236:51723 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP wh1.24dial.com:5060;rport;received=23.251.129.236;branch=z9hG4bK123456789
Call-ID: 4a75f291-3b82-4920-ad1e-2d43360ebeff
From: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com>;tag=123456789
To: <sip:018053557555@213.71.23.139>;tag=ae356e6b-e90e-457d-9c17-8689f75eafd9
CSeq: 24500 BYE
Content-Length:  0





On the other hand when Asterisk responds to a SIP/OK with an SIP/ACK it sends the message to the correct port:

<--- Transmitting SIP request (1421 bytes) to UDP:23.251.129.236:5060 --->
INVITE sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com:5060 SIP/2.0
Via: SIP/2.0/UDP 213.71.23.139:5060;rport;branch=z9hG4bKPj0bfc26d4-8b6c-46bb-8a65-e936430e86b9
From: <sip:018053557555@213.71.23.139>;tag=ae356e6b-e90e-457d-9c17-8689f75eafd9
To: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com>
Contact: <sip:3e315314-9519-450e-a0fe-45912f493819@213.71.23.139:5060>
Call-ID: 4a75f291-3b82-4920-ad1e-2d43360ebeff
CSeq: 24500 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   745


<--- Received SIP response (1250 bytes) from UDP:23.251.129.236:35623 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.71.23.139:5060;rport;branch=z9hG4bKPj0bfc26d4-8b6c-46bb-8a65-e936430e86b9;rport=5060;received=213.71.23.139
From:  <sip:018053557555@213.71.23.139>;tag=ae356e6b-e90e-457d-9c17-8689f75eafd9
To:  <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com>;tag=123456789
Contact: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com:5060>
Call-ID: 4a75f291-3b82-4920-ad1e-2d43360ebeff
CSeq: 24500 INVITE
Content-Type: application/sdp
Content-Length: 780


<--- Transmitting SIP request (384 bytes) to UDP:23.251.129.236:5060 --->
ACK sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com:5060 SIP/2.0
Via: SIP/2.0/UDP 213.71.23.139:5060;rport;branch=z9hG4bKPjcde83bb3-4750-4869-96db-692806b049d5
From: <sip:018053557555@213.71.23.139>;tag=ae356e6b-e90e-457d-9c17-8689f75eafd9
To: <sip:evj6xZZffPGL8gOmnx0KL@wh1.24dial.com>;tag=123456789
Call-ID: 4a75f291-3b82-4920-ad1e-2d43360ebeff
CSeq: 24500 ACK
Content-Length:  0


Any ideas, what might be the problem?
Comments:By: Matt Jordan (mjordan) 2014-01-24 06:23:14.350-0600

Can you test with the latest from the Asterisk 12 branch?

A number of NAT issues have been fixed in {{res_pjsip_nat}}, including some issues where a few messages were sent to the wrong destination.

If that doesn't resolve your issue, please attach your {{pjsip.conf}}.
By: Marko Seidenglanz (markose) 2014-01-24 06:28:09.489-0600

Ok. I will try the latest version from repo first. Thanks for the advice!
By: Rusty Newton (rnewton) 2014-02-10 08:05:12.225-0600

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines