Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
Alex
Hi Alex!
This is done in outside the proxy - for in-dialog requests, the request URI contains the "contact" of the other party (or the Route header for strict routers (old SIP RFC))
regards, klaus
Alex Bligh wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
Alex
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--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
Alex
On Wednesday 02 June 2004 20:03, Alex Bligh wrote:
--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
In this case the Route header resulting out of the Record-Route header includes all informations required to route any subsequent in-dialog request to it's destination. So any re-INVITE or BYE can be routed stateless to its destination.
Greetings Nils
Nils / Klaus,
--On 02 June 2004 20:08 +0200 Nils Ohlmeier nils@iptel.org wrote:
On Wednesday 02 June 2004 20:03, Alex Bligh wrote:
--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
Nils said:
In this case the Route header resulting out of the Record-Route header includes all informations required to route any subsequent in-dialog request to it's destination. So any re-INVITE or BYE can be routed stateless to its destination.
Klaus said:
This is done in outside the proxy - for in-dialog requests, the request URI contains the "contact" of the other party (or the Route header for strict routers (old SIP RFC))
This implies ser is looking either at the parameters in the Record-Route: header added by ser on the INVITE (which is what Nils seems to suggest) OR relies on the initiating UA using as a Request-URI for subsequent in-dialog messages the Contact: header added by the UA providing the OK response to INVITE.
I think it's probably the latter that's happening. But can I rely on this? IE Is it mandatory for the subsequent in-dialog transactions to use the Contact: address supplied by the UAS, and is it mandatory for the UAS to add a Contact: header at all? I'm reading the call flow example on p167 of "SIP: Understanding the Session Initiation Protocol" and it doesn't seem to be there. What's concerning me is that I can see it works, but I can't tell why or if it will always work. I am probably being very dumb.
Alex
Hi Alex,
On Wednesday 02 June 2004 22:26, Alex Bligh wrote:
Nils / Klaus,
--On 02 June 2004 20:08 +0200 Nils Ohlmeier nils@iptel.org wrote:
On Wednesday 02 June 2004 20:03, Alex Bligh wrote:
--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
Nils said:
In this case the Route header resulting out of the Record-Route header includes all informations required to route any subsequent in-dialog request to it's destination. So any re-INVITE or BYE can be routed stateless to its destination.
Klaus said:
This is done in outside the proxy - for in-dialog requests, the request URI contains the "contact" of the other party (or the Route header for strict routers (old SIP RFC))
This implies ser is looking either at the parameters in the Record-Route: header added by ser on the INVITE (which is what Nils seems to suggest) OR relies on the initiating UA using as a Request-URI for subsequent in-dialog messages the Contact: header added by the UA providing the OK response to INVITE.
I think it's probably the latter that's happening. But can I rely on this? IE Is it mandatory for the subsequent in-dialog transactions to use the Contact: address supplied by the UAS, and is it mandatory for the UAS to add a Contact: header at all? I'm reading the call flow example on p167 of "SIP: Understanding the Session Initiation Protocol" and it doesn't seem to be there. What's concerning me is that I can see it works, but I can't tell why or if it will always work. I am probably being very dumb.
As I allready explained on SIP-Implemetors: the Contact header is mandatory for requests and replies which can establish a dialog. If your book does not mention this or even do not explain the simple scenario how further in-dialog requests (re-INVITE, BYE's etc.) are delivered to the target, i would suggest to throw it away and read the RFC instead ;-)
The Contact from the remote side will be used in future requests as final target. In case of loose-routing as request URI, in case of strict routing as last hop in the Route header. That is the reason why the Contact header is mandatory in INVITE and 200 OK (for the INVITE). (That is the reason why i said in the Route header is everything included.) So the proxy will never rely on parameters in the Record-Route/Route header.
If you are looking at some real world traces, search for the remote side Contact URI in the BYE message. You will for sure find it somewhere.
Greetings Nils
Alex Bligh wrote:
Nils / Klaus,
--On 02 June 2004 20:08 +0200 Nils Ohlmeier nils@iptel.org wrote:
On Wednesday 02 June 2004 20:03, Alex Bligh wrote:
--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
Nils said:
In this case the Route header resulting out of the Record-Route header includes all informations required to route any subsequent in-dialog request to it's destination. So any re-INVITE or BYE can be routed stateless to its destination.
Klaus said:
This is done in outside the proxy - for in-dialog requests, the request URI contains the "contact" of the other party (or the Route header for strict routers (old SIP RFC))
This implies ser is looking either at the parameters in the Record-Route: header added by ser on the INVITE (which is what Nils seems to suggest) OR relies on the initiating UA using as a Request-URI for subsequent in-dialog messages the Contact: header added by the UA providing the OK response to INVITE.
I think it's probably the latter that's happening. But can I rely on this? IE Is it mandatory for the subsequent in-dialog transactions to use the Contact: address supplied by the UAS, and is it mandatory for the UAS to add a Contact: header at all? I'm reading the call flow example on p167 of "SIP: Understanding the Session Initiation Protocol" and it doesn't seem to be there. What's concerning me is that I can see it works, but I can't tell why or if it will always work. I am probably being very dumb.
you have to distinguish between loose router (new RFC) and strict routers. For both, a dialog creating response typically (not always mandatory) includes the Contact: header.
- loose routers: the Contact:-URI is in the request URI and the Route: header indicates the intermediate SIP proxies.
- strict_routers: the next hop is in the request URI and the Contact:-Uri is in the last Route: header
Anyway, the information to route the message to the proper UA is available.
Klaus
At 10:26 PM 6/2/2004, Alex Bligh wrote:
Nils / Klaus,
--On 02 June 2004 20:08 +0200 Nils Ohlmeier nils@iptel.org wrote:
On Wednesday 02 June 2004 20:03, Alex Bligh wrote:
--On 02 June 2004 18:38 +0100 Alex Bligh alex@alex.org.uk wrote:
Consider the situation whereby after an INVITE occurs which ser initially routes to A (which is, say, busy) and subsequently routes to B. Perhaps it has gone through t_on_failure or some other mechanism of sequential forking. Assume the INVITE has been OK'd and ACK'd.
What mechanism does ser use to ensure subsequent transactions on the call-leg (such as BYE, re-INVITE etc.) are sent to B and not A? IE where is the state being stored and how?
I am obviously assuming (transaction) stateful forwarding and that record_route is on (else the proxy wouldn't even get to see BYE etc.)
Nils said:
In this case the Route header resulting out of the Record-Route header includes all informations required to route any subsequent in-dialog request to it's destination. So any re-INVITE or BYE can be routed stateless to its destination.
Klaus said:
This is done in outside the proxy - for in-dialog requests, the request URI contains the "contact" of the other party (or the Route header for strict routers (old SIP RFC))
This implies ser is looking either at the parameters in the Record-Route: header added by ser on the INVITE (which is what Nils seems to suggest) OR relies on the initiating UA using as a Request-URI for subsequent in-dialog messages the Contact: header added by the UA providing the OK response to INVITE.
In terms of RFC3261, it routes by r-uri, unless the next hop is driven by Route header field.
I think it's probably the latter that's happening. But can I rely on this? IE Is it mandatory for the subsequent in-dialog transactions to use the Contact: address supplied by the UAS, and is it mandatory for the UAS to add a Contact: header at all? I'm reading the call flow example on p167 of "SIP: Understanding the Session Initiation Protocol" and it doesn't seem to be there. What's concerning me is that I can see it works, but I can't tell why or if it will always work. I am probably being very dumb.
In RFC3261, use of Contacts is mandatory. That's not the case in RFC2543 which was a bug.
-jiri
--On 03 June 2004 01:15 +0200 Jiri Kuthan jiri@iptel.org wrote:
In RFC3261, use of Contacts is mandatory. That's not the case in RFC2543 which was a bug.
Ah! the final part of the jigsaw. The SIP book (written by an RFC author) seems to predate RFC3261 (at least in parts). Thanks. Now all sorted I think.
Alex
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Alex Bligh -
Jiri Kuthan -
Klaus Darilion -
Nils Ohlmeier