Jeroen,

Thanks a lot for your emails. Let us say the call distributor hands off these 
INVITEs to UAs (and not proxies), then the UA can directly send the response to 
the UAC, say 180 or 200 with its own Contact header. In such a case, the PRACK 
(for 180) or ACK for 200 Ok, would directly reach the UA. A re-transmitted 
INVITEs would also get routed to same UA (until the distributor maintains the 
state). Would such an approach work for TCP as well? Or in cases of TCP the ACK 
would still have to go thorugh the distributor and later the UAS should send an 
UPDATE to change the port?

If some clients do use same call-id, but different From tags, they would end up 
being routed to the same UA which would maintain full state and can distinguish 
them as seperate calls.

By looking at just the call-id, the distributor need not do a full SIp decoding 
and also need not maintain too much state.

Thanks

Andy



Jeroen van Bemmel <[EMAIL PROTECTED]> wrote:
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" 

> To: "Paul Kyzivat" 

> Cc: 
> 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 
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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