[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 11:48 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 11:48  
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as seen from ASTERISK: 

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK130df927
From: "7059998" <sip:7059998 at 192.168.1.28>;tag=as31aa884a
To: <sip:34669448337 at 192.168.1.18>;tag=00E0F51004DB3090910500000F87
Call-ID: 077801314ac978605279977268fdf0b6 at 192.168.1.28
CSeq: 102 INVITE
Contact: <sip:192.168.1.18:5060>       <----- CONTACT
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:669448337 at 192.168.1.18:3111>
Content-Length:   200

v=0
o=- 81478068100004999 3 IN IP4 192.168.1.18
s=session
c=IN IP4 192.168.1.18
t=0 0
m=audio 5500 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:40
a=sendrecv
a=rtpmap:101 telephone-event/8000

as seen from TSHARK:

#
U 2008/01/21 18:39:59.675875 192.168.1.18:3111 -> 192.168.1.28:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK130df927
From: "7059998" <sip:7059998 at 192.168.1.28>;tag=as31aa884a
To: <sip:34669448337 at 192.168.1.18>;tag=00E0F51004DB3090910500000F87
Call-ID: 077801314ac978605279977268fdf0b6 at 192.168.1.28
CSeq: 102 INVITE
Contact: <sip:192.168.1.18:5060>
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:669448337 at 192.168.1.18>
Content-Length:   200

v=0
o=- 81478068100004999 3 IN IP4 192.168.1.18
s=session
c=IN IP4 192.168.1.20
t=0 0
m=audio 5500 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:40
a=sendrecv
a=rtpmap:101 telephone-event/8000


This is too weird... 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-21-08 11:48  davidcsi       Note Added: 0080952                          
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