[asterisk-bugs] [Asterisk 0016122]: RTP Media Port Change Ignored
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Dec 10 14:13:07 CST 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16122
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Reported By: teamforrest
Assigned To: jpeeler
======================================================================
Project: Asterisk
Issue ID: 16122
Category: Channels/chan_sip/General
Reproducibility: have not tried
Severity: minor
Priority: normal
Status: assigned
Target Version: 1.6.0.21
Asterisk Version: SVN
JIRA: SWP-401
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-10-23 14:51 CDT
Last Modified: 2009-12-10 14:13 CST
======================================================================
Summary: RTP Media Port Change Ignored
Description:
In a SIP conversation, asterisk is sending an OK to a SIP Reinvite with a
media port change. Asterisk is not using this new port and will continue to
use the original port in the original invite. In the example below, when
connecting to 71.116.126.81, the original media port was 7750. A reinvite
was made to port 7752, however asterisk will continue to use 7750. This can
be reproduced and has same behavior no matter how the canreinvite flag is
set (ie no, yes, update, etc.). This can be tested by calling to
3366 at conf.zipdx.com and pressing 2. This test was done to
3366 at conf.hificonf.com, which is a lab server active at the time of this
post.
NGREP of SIP traffic:
ngrep -W byline -q -T 71.116.126.81
interface: eth0 (172.16.1.0/255.255.255.0)
match: 71.116.126.81
U +10.175362 71.116.126.81:5060 -> 172.16.1.220:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 76.206.40.250:5060;rport=1024;branch=z9hG4bK05f69878.
From: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 102 INVITE.
To: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
Content-Length: 0.
Contact: <sip:71.116.126.81;transport=udp>.
Server: ZipDX-3.10.3.
.
U +0.030072 71.116.126.81:5060 -> 172.16.1.220:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 76.206.40.250:5060;rport=1024;branch=z9hG4bK05f69878.
From: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 102 INVITE.
To: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
Content-Length: 203.
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS, INFO, REGISTER,
SUBSCRIBE, MESSAGE.
Contact: <sip:71.116.126.81;transport=udp>.
Server: ZipDX-3.10.3.
Supported: timer.
.
v=0.
o=telStage 270 3465312789 IN IP4 71.116.126.81.
s=-.
c=IN IP4 71.116.126.81.
t=0 0.
m=audio 7750 RTP/AVP 9 101.
a=rtpmap:9 G722/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
U +0.000307 172.16.1.220:5060 -> 71.116.126.81:5060
ACK sip:71.116.126.81;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 76.206.40.250:5060;branch=z9hG4bK2b902523;rport.
Max-Forwards: 70.
From: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
To: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
Contact: <sip:qxork at 76.206.40.250>.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.0.15.
Content-Length: 0.
.
U +11.995979 71.116.126.81:5060 -> 172.16.1.220:5060
INVITE sip:qxork at 76.206.40.250 SIP/2.0.
From: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
To: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Contact: <sip:71.116.126.81;transport=udp>.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 20447 INVITE.
Content-Length: 184.
Content-Type: application/sdp.
Supported: timer.
User-Agent: ZipDX-3.10.3.
Max-Forwards: 70.
Via: SIP/2.0/UDP
71.116.126.81:5060;branch=z9hG4bK4ae1fba1-0050-00004858-6a76558a-8d23bf0c.
.
v=0.
o=telStage 271 3465312801 IN IP4 71.116.126.81.
s=-.
c=IN IP4 71.116.126.81.
t=0 0.
m=audio 7752 RTP/AVP 9 96.
a=rtpmap:9 G722/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
U +0.000335 172.16.1.220:5060 -> 71.116.126.81:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
71.116.126.81:5060;branch=z9hG4bK4ae1fba1-0050-00004858-6a76558a-8d23bf0c;received=71.116.126.81.
From: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
To: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 20447 INVITE.
User-Agent: Asterisk PBX 1.6.0.15.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Contact: <sip:qxork at 76.206.40.250>.
Content-Length: 0.
.
U +0.000041 172.16.1.220:5060 -> 71.116.126.81:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
71.116.126.81:5060;branch=z9hG4bK4ae1fba1-0050-00004858-6a76558a-8d23bf0c;received=71.116.126.81.
From: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
To: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 20447 INVITE.
User-Agent: Asterisk PBX 1.6.0.15.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Contact: <sip:qxork at 76.206.40.250>.
Content-Type: application/sdp.
Content-Length: 263.
.
v=0.
o=root 1586015446 1586015447 IN IP4 76.206.40.250.
s=Asterisk PBX 1.6.0.15.
c=IN IP4 76.206.40.250.
t=0 0.
m=audio 10378 RTP/AVP 9 96.
a=rtpmap:9 G722/8000.
a=rtpmap:96 telephone-event/8000.
a=fmtp:96 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
U +0.118024 71.116.126.81:5060 -> 172.16.1.220:5060
ACK sip:qxork at 76.206.40.250 SIP/2.0.
From: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
To: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
Contact: <sip:71.116.126.81;transport=udp>.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 20447 ACK.
Content-Length: 0.
Via: SIP/2.0/UDP
71.116.126.81:5060;branch=z9hG4bK4ae1fba1-00dd-00004859-6a76558a-8d23bf0c.
.
U +15.023183 172.16.1.220:5060 -> 71.116.126.81:5060
BYE sip:71.116.126.81;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 76.206.40.250:5060;branch=z9hG4bK519c988b;rport.
Max-Forwards: 70.
From: "Fred Posner" <sip:qxork at 76.206.40.250>;tag=as61320760.
To: <sip:3366 at conf.hificonf.com>;tag=telStage-2d24-4ae1fb95.
Call-ID: 723341d540ae29472b91211466cf500d at 76.206.40.250.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 1.6.0.15.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.
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----------------------------------------------------------------------
(0115093) lmadsen (administrator) - 2009-12-10 14:13
https://issues.asterisk.org/view.php?id=16122#c115093
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If it is expected behaviour, then I believe some sort of documentation
update need to be done to make this clear to the end user.
However, here is a conversation on IRC in #vuc which makes me wonder why
1.6.2.0 acts differently than 1.6.0 and 1.6.1:
<DocAwesome> qxork: when you enable nat=yes in 1.6.2.0, does it still
work?
<qxork> yes
<DocAwesome> hmmmmmmmmmm
<qxork> I know
<DocAwesome> that makes me wonder if there is something broken in 1.6.2.0
<DocAwesome> maybe it SHOULDN'T work :)
<qxork> I tried it with nat=yes and nat=no
<qxork> in 1.6.1 or 1.6.0
<qxork> with same results.
<DocAwesome> my gut instinct is that it has something to do with the
kill-the-user stuff...
<DocAwesome> qxork: you're saying nat=yes in 1.6.0 and 1.6.1 causes it to
drop audio, but not in 1.6.2.0 ?
<DocAwesome> or that you don't get dropped audio at all anymore?
<qxork> in 1.6.0 and 1.6.1 I dropped audio with nat=yes. I also dropped it
with nat=no
<qxork> in 1,6,2 I have nat=yes and have not dropped
<DocAwesome> huh... that's even more strange as I don't have the issue
with dropped audio with 1.6.1.10 with nat=no
<DocAwesome> something is not in sync then
Issue History
Date Modified Username Field Change
======================================================================
2009-12-10 14:13 lmadsen Note Added: 0115093
======================================================================
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