On second thought, you'd better include the from-tag too (if present), it
adds some extra protection against buggy clients that use not-so-random
call-ids (though not foolproof).
If your proxies are such that they set to-tags by which the proxy instance
can be recognised, you can also consider to use that as primary or fallback
routing mechanism for subsequent requests. Requests after the first INVITE
(say PRACK) will contain such a to-tag identifying the proxy that initially
answered. In addition, you can cache the to-tag you find in a response to
route subsequent requests (without a need for the identifying aspect of
to-tags, but drawback that it means you need to keep state at the
distributor)
Jeroen
Hi Andy,
In theory I would say: yes, Call-Id would be sufficient
The complete dialog-id consists of from-tag + call-id + to-tag, the to-tag
is not set yet and the caller could send multiple INVITEs with the same
call-id but different from tags. Consequence would be that all these calls
get routed to the same proxy, but that seems reasonable to me
An issue that will occur is: at which point will you remove the state in
the distributor (i.e. the mapping of call-id to the selected proxy)?
Perhaps upon reception of a response containing a Contact header, but then
what about retransmitted INVITEs / PRACKS etc that arrive after that? You
would probably need to set a timer and wait for 32 seconds after
forwarding the response with the contact header
Jeroen
----- Original Message -----
From: "Andy Pandaram" <[EMAIL PROTECTED]>
To: "Paul Kyzivat" <[EMAIL PROTECTED]>
Cc: <[email protected]>
Sent: Thursday, June 30, 2005 8:28 AM
Subject: Re: [Sip-implementors] Using call-id for SIP call distribution
The call distributor only distributes incoming SIP calls among multiple
proxies or may be even SIP UAs (like a SIP-PSTN GW). The SIP-GWs would
listen on non-standard ports. The call distributor would receive INVITEs
etc on port 5060 and send them to the SIP GW which may be listening on
port 10,000. Now, the re-transmitted INVITEs would still come to the call
distributor. And also until the SIP GW can give its new contact in the
200 Ok or possibly in an UPDATE, subsequent requets (like PRACK) might
come through the call distributor. In such cases, the distributor would
need to send the subsequent requests to the same GW (as the first INVITE
was sent to). Now, for this purpose, is it enough that the distributor
looks at just the Call-Id and not other tags and parameters?
Thanks
Andy
Paul Kyzivat <[EMAIL PROTECTED]> wrote:
Andy Pandaram wrote:
Hi,
If a SIP call distributor has to send incoming calls to multiple SIP
proxies, is it enough to just look at the Call-Id (since that must be
unique across space and time)? Is there any reason for the distributor
to look at From/To Tags/Via branch etc?
What is a SIP call distributor?
If it is a proxy then 3261 should tell you all you need to know.
Paul
Thanks
Andy
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