[Serusers] RFC3261 Compliance???

[Java Rockx]

I may have stumbled upon the root of the problem and a solution.

In my ser.cfg I have

if (!method=="INVITE") record-route();

And my phone is a Grandstream BT100. I did not have anything in my "Outbound Proxy" field because
I assumed that omitting this field would default the phone to use the ser proxy as the outbound
proxy.

I now see that this is not the case. When my outbound proxy field was left blank my ACK looked
like this:

NOTE: These ACKs are originating at my ser proxy and being sent to the Sonus equipment.

U 2004/11/18 02:19:05.789518 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:216.12.18.98:5060;lr SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP 172.16.1.34;rport=5060;received=66.90.50.230;branch=z9hG4bK0cf6f59ad9cff263.
From: "Paul Hazlett" <sip:9990010005 at sip.mycompany.com;user=phone>;tag=153be9cfe0227ba6.
To: <sip:14075551212 at sip.mycompany.com;user=phone>;tag=0bc2b91b.
Contact: <sip:9990010005 at 66.90.50.230:5060;user=phone>.
Proxy-Authorization: DIGEST username="9990010005", realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212 at 216.12.18.98:5060", nonce="419c4e0f62b492f993917bb283d89431c978b522",
response="4dd7cb5d0e06a50c24d3a4848f57f1ad".
Call-ID: 2fe29b9654abc94f at 172.16.1.34.
CSeq: 16138 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.



Now with my outbound proxy field set to the same value as my sip proxy I get this ACK -- notice
the 
Route: header??? The Sonus equipment now seems to be happy.

U 2004/11/18 02:29:47.505321 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:4075551212 at 216.229.118.76:4060 SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP 66.90.50.230;branch=z9hG4bKd14abbdd801339a6.
Route: <sip:216.12.18.98:5060;lr>.
From: "Paul Hazlett" <sip:9990010005 at sip.mycompany.com;user=phone>;tag=343704cb47bd81b2.
To: <sip:14075551212 at sip.mycompany.com;user=phone>;tag=06c57864.
Contact: <sip:9990010005 at 66.90.50.230;user=phone>.
Proxy-Authorization: DIGEST username="9990010005", realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212 at 216.229.118.76:4060", nonce="419c50915314f17ec7beef27579c4fc2ccfba7ed",
response="6cde7461dfde9e0a7327fa1a20f47fb0".
Call-ID: f211021ad0a34a37 at 172.16.1.34.
CSeq: 3685 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.




--- Java Rockx <javarockx at yahoo.com> wrote:

> Ranga,
> 
> Yes, in my ser.cfg script I have
> 
> if (!method=="REGISTER") record-route();
> 
> I think there is plenty of confusion in the serusers list about wheather or not this is correct.
> Most of the example ser.cfg files I've seen, including those that sip with ser, show this rather
> than
> 
> if (!method=="INVITE") record-route();
> 
> Can anyone say which method is more RFC3261 compliant? 
> 
> Also, I can't say wheather or not changing my ser.cfg to the latter will fix the problems that
> our
> PSTN provider has with our SIP dialogs. I'll find that out in the morning.
> 
> Regards,
> Paul
> 
> 
> --- Ranga <rangarao.v at gmail.com> wrote:
> 
> > > U 2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
> > > ACK sip:4075551212 at 216.229.127.60:5060 SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
> > > Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
> > > Route: <sip:216.229.127.60:5060;lr>.
> > > From: "Paul (1002)" <sip:9990010001 at sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
> > > To: <sip:14075551212 at sip.mycompany.com;user=phone>;tag=0bf5212d.
> > > Contact: <sip:9990010001 at 192.168.0.83;user=phone>.
> > > Proxy-Authorization: DIGEST username="9990010001", realm="sip.mycompany.com", algorithm=MD5,
> > uri="sip:4075551212 at 216.229.127.60:5060", nonce="419baee2139477c22db6d0ec9a47b23003512431",
> > response="7b83435f78bc60f5ced80304c7d54093".
> > > Call-ID: aa31202374f54793 at 192.168.0.83.
> > > CSeq: 12010 ACK.
> > > User-Agent: Grandstream BT100 1.0.5.11.
> > > Max-Forwards: 70.
> > > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> > > Content-Length: 0.
> > > .
> > > #
> > > U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
> > > ACK sip:216.229.127.60:5060;lr SIP/2.0.
> > > Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
> > > Via: SIP/2.0/UDP 68.80.201.101;branch=0.
> > > Via: SIP/2.0/UDP
> > 192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
> > > From: "Paul (1002)" <sip:9990010001 at sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
> > > To: <sip:14075551212 at sip.mycompany.com;user=phone>;tag=0bf5212d.
> > > Contact: <sip:9990010001 at 68.80.200.100:1063;user=phone>.
> > > Proxy-Authorization: DIGEST username="9990010001", realm="sip.mycompany.com", algorithm=MD5,
> > uri="sip:4075551212 at 216.229.127.60:5060", nonce="419baee2139477c22db6d0ec9a47b23003512431",
> > response="7b83435f78bc60f5ced80304c7d54093".
> > > Call-ID: aa31202374f54793 at 192.168.0.83.
> > > CSeq: 12010 ACK.
> > > User-Agent: Grandstream BT100 1.0.5.11.
> > > Max-Forwards: 16.
> > > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> > > Content-Length: 0.
> > > .
> > > #
> > 
> > I guess, you are doing record-route for ACK also. IMO, you should not
> > add record route headers for ACK.
> > 
> > I am not sure why Route header is completely removed from the request.
> > Proxy is supposed to delete only the top route value.
> > 
> > -Ranga
> > 
> 
> 
> 
> 		
> __________________________________ 
> Do you Yahoo!? 
> The all-new My Yahoo! - Get yours free! 
> http://my.yahoo.com 
>  
> 
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
> 



		
__________________________________ 
Do you Yahoo!? 
Meet the all-new My Yahoo! - Try it today! 
http://my.yahoo.com