[asterisk-dev] canreinvite=no behaviour changed between 1.2.xand1.4 ?

Wed Oct 11 12:45:26 MST 2006

Olle, Luigi, 

Look at the To and the From in this re-invite transaction in v1.4 (got from
v1.4 svn  today, been going on since beta 2 release though) they're flipped
in the ACK to the 200 OK. They're fine before that.
XXX's to protect the innocent. Scratching my head on this one. Only happens
on a re-invite not the initial call setup.

Start of reinvite 

U XXX.4.233.235:5060 -> XXX.242.225.200:5060
INVITE sip:1XXX6532537 at XXX.242.225.200:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP XXX.4.233.235:5060;branch=z9hG4bK48402b67;rport.
From: <sip:XXX3680400 at XXX.242.225.200;user=phone>;tag=as1f372c0a.
To: OUT_OF_AREA
<sip:1XXX6532537 at XXX.242.225.158;user=phone>;tag=SD15ktc01-1160588810-665744
1161254554-11.
Contact: <sip:XXX3680400 at XXX.4.233.235>.
Call-ID: SD15ktc01-c6d1545951635356cfcf6b10f730e419-vjvtfv3.
CSeq: 102 INVITE.
User-Agent: Asterisk.
Max-Forwards: 70.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 246.
.
v=0.
o=root 30886 30887 IN IP4 XXX.242.225.200.
s=session.
c=IN IP4 XXX.242.225.200.
t=0 0.
m=audio 16412 RTP/AVP 0 127.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:127 telephone-event/8000.
a=fmtp:127 0-16.
a=silenceSupp:off - - - -.
a=sendrecv.

U XXX.242.225.200:5060 -> XXX.4.233.235:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP XXX.4.233.235:5060;branch=z9hG4bK48402b67;rport=5060.
From: <sip:XXX3680400 at XXX.242.225.200;user=phone>;tag=as1f372c0a.
To: OUT_OF_AREA
<sip:1XXX6532537 at 127.242.225.158;user=phone>;tag=SD15ktc01-1160588810-665744
1161254554-11.
Call-ID: SD15ktc01-c6d1545951635356cfcf6b10f730e419-vjvtfv3.
CSeq: 102 INVITE.

SIP/2.0 200 OK.
Via: SIP/2.0/UDP XXX.4.233.235:5060;branch=z9hG4bK48402b67;rport=5060.
From: <sip:XXX3680400 at XXX.242.225.200;user=phone>;tag=as1f372c0a.
To: OUT_OF_AREA
<sip:1XXX6532537 at XXX.242.225.158;user=phone>;tag=SD15ktc01-1160588810-665744
1161254554-11.
Call-ID: SD15ktc01-c6d1545951635356cfcf6b10f730e419-vjvtfv3.
CSeq: 102 INVITE.
Contact: <sip:1XXX6532537 at XXX.242.225.200:5060;transport=udp>.
Accept-Language: en; q=0.0.
Content-Type: application/sdp.
Content-Length: 266.
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY,
PRACK, UPDATE.
.
v=0.
o=Sonus_UAC 4821 2801 IN IP4 XXX.242.225.200.
s=SIP Media Capabilities.
c=IN IP4 XXX.242.225.200.
t=0 0.
m=audio 16410 RTP/AVP 18 0 127.
a=rtpmap:18 G729/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:127 telephone-event/8000.
a=fmtp:127 0-15.
a=sendrecv.
a=maxptime:20.

#
U XXX.4.233.235:5060 -> XXX.242.225.200:5060
ACK sip:1XXX6532537 at XXX.242.225.200:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP XXX.4.233.235:5060;branch=z9hG4bK1396ae1c;rport.
From: OUT_OF_AREA
<sip:1XXX6532537 at XXX.242.225.158;user=phone>;tag=SD15ktc01-1160588810-665744
1161254554-11.
To: <sip:XXX3680400 at XXX.242.225.200;user=phone>;tag=as1f372c0a.
Contact: <sip:XXX3680400 at XXX.4.233.235>.
Call-ID: SD15ktc01-c6d1545951635356cfcf6b10f730e419-vjvtfv3.
CSeq: 102 ACK.
User-Agent: Asterisk.
Max-Forwards: 70.
Content-Length: 0. 

Kevin Collins

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Kevin Collins
Sent: Wednesday, October 11, 2006 12:27 PM
To: 'Asterisk Developers Mailing List'
Subject: RE: [asterisk-dev] canreinvite=no behaviour changed between
1.2.xand1.4 ?

Olle,

I'm seeing some strange re-invite behaviors as well in 1.4. I saw similar
issues as Luigi did as well.

I'll update later with examples. 

Kevin Collins

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Johansson Olle E
Sent: Wednesday, October 11, 2006 1:56 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] canreinvite=no behaviour changed between 1.2.x
and1.4 ?

11 okt 2006 kl. 03.22 skrev Kevin P. Fleming:

> ----- Luigi Rizzo <rizzo at icir.org> wrote:
>> Quite a bit of investigations (full description below) made me think 
>> that on 1.4 the "canreinvite=no" is honored only for calls coming 
>> from the device to asterisk, whereas on 1.2.x it was used in both 
>> directions.
>> So i wonder, do i understand well, and is the change intentional ?
>> It doesn't make any sense to me to differentiate the behaviour basing 
>> on the originator of the call, because you still have media flowing 
>> in both directions in both cases.
>
> No, the canreinvite setting should apply to both users and peers 
> equally.
>

And for both incoming and outbound calls for a peer :-)

/O

---
Olle E. Johansson * Asterisk Evangelist, developer * VOOP A/S olle at voop.com

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