15 Feb
2004
15 Feb
'04
9:46 p.m.
Hi,
I do have the following situation:
UA1 calls UA2 going through ser 0.8.12. All signaling is Recourd Routed and
go through ser.
UA2 doesn't answer and I do have a t_on_failure to go to UA3.
UA3 receives INVITES from ser, and answer with a 200/OK with sdp, this
messages goes through ser which forward it to UA1.
UA1 sends an ACK to ser having this in message:
Session Initiation Protocol
Request line: ACK sip:ua3@22.2.4.193:5071 SIP/2.0
Method: ACK
Message Header
Via: SIP/2.0/UDP 2.20.64.193:5063;branch=z9hG4bk-7f415771
From: test1 <sip:ua1@vocal.ipsound.net>;tag=eb646133bf338b12
To: <sip:ua2@vocal.ipsound.net>;tag=as47221fa0
Call-ID: a66804e6-5ac924fd(a)142.133.80.102
CSeq: 102 ACK
Max-Forwards: 70
Route: <sip:sip-proxy@ser-domain.com;ftag=eb646133bf338b12;lr=on>
Proxy-Authorization: Digest username="
Contact: ua1 <sip:ua1@24.202.64.193:5063>
User-Agent: Sipura/SPA2000-1.0.10
Content-Length: 0
Ser is supposed to follow the route, and ends up looking into the location
table and re-forwards the request to UA2 again (even if it already timed out
once).
Now I do have another UA which in this same scenario sends the correct ACK:
Session Initiation Protocol
Request line: ACK sip:sip-proxy@ser-domain.com SIP/2.0
Method: ACK
Message Header
Route: <sip:ua3@22.2.4.193:5071>
Via: SIP/2.0/UDP 2.20.64.193:5063
From: ua1 <sip:ua1@ser-domain.com;user=phone>;tag=3551608275
To: <sip:ua2@ser-domain.com;user=phone>;tag=as112d04ea
Call-ID: 2653016335(a)142.133.80.101
CSeq: 2 ACK
User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A)
Proxy-Authorization: Digest username
Content-Length: 0
In this case ser follow the route header, and ends up sending the ACK to
UA3.
Snom phone with 2.03o firmware as well as sipura with 1.0.10 and 1.0.29b
firmware have the same wrong behaviour.
What is the right way of having this ?
My understanding is that in record route situation, ser looks into the route
header, and sends the message based on it. Here I do have User Agents which
seem to create an ACK message ignoring it should go to a sip proxy .
Am I missing something here ?
Thanks.
Samy.