[asterisk-bugs] [Asterisk 0011710]: RTPs not sent to the correct IP

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11710 
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Reported By:                davidcsi
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11710
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.16.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-08-2008 15:25 CST
Last Modified:              01-21-2008 11:06 CST
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Summary:                    RTPs not sent to the correct IP
Description: 
When outgoing call to a gateway with 2 IPs, 1 for signalling and 1 or more
for RTPs, asterisk sends the RTPs to the signalling IP ignoring the media
connection IP.

The session progress says its RTP IP is .20, so does Ringing. Though on
200 OK it says it is .18

Why would Asterisk change the RTP IP because of an ackwoledge?


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---------------------------------------------------------------------- 
 davidcsi - 01-21-08 11:06  
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There's no proxy involved, this is a ngrep executed on the same box:

#
U 2008/01/21 18:08:01.368095 80.65.12.18:1382 -> 80.65.12.28:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 80.65.12.28:5060;branch=z9hG4bK10683fff
From: "7059998" <sip:niai7059998 at 80.65.12.28>;tag=as2baccc6a
To:
<sip:M3ll4m0d4v1d13058883456 at 80.65.12.18>;tag=00E0F51004DB308DA5D9000010BB
Call-ID: 51397766734aac29320023413ebf4e95 at 80.65.12.28
CSeq: 102 INVITE
Contact: <sip:80.65.12.18:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Supported: 100rel, timer, replaces
P-Asserted-Identity: <sip:80.65.12.18>
Privacy: id
Content-Length:   200

v=0
o=- 81458940600001582 2 IN IP4 80.65.12.18
s=session
c=IN IP4 80.65.12.20
t=0 0
m=audio 4340 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:40
a=sendrecv
a=rtpmap:101 telephone-event/8000

As yu can see, the packet is coming in right. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-21-08 11:06  davidcsi       Note Added: 0080946                          
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