484 seems to me, a very bad way of doing overlapped dialling in SIP.
 
 
 
 
 
 
 
 
rfc 3578 (http://www.ietf.org/rfc/rfc3578.txt)  says:
 
   Routing in SIP can be controlled by the administrator of the network.
   Therefore, a gateway can be configured to generate SIP overlap
   signalling in the way described below only if the SIP routing
   infrastructure ensures that INVITEs will only reach one gateway.

 
What if you are using something like DNS SRV to route to a group
of gateways?   How can one ensure all new requests go the same gateway?
 
RFC 3578 then says
 
   When the routing infrastructure is not under the control of the
   administrator of the gateway, the procedures of Section 2 have to be
   used instead.

 
The problem is that section 2 talks about "Waiting for the Minimum
Amount of Digits"
but this is not very useful - effectively it stops you using overlap
dialling.
 
 
Now, the RFC discourages use of the route learnt in 484:
 
      Note that it is possible to receive a response to the first INVITE
      before having sent the second INVITE.  In this case, the response
      received would contain a To tag and information (Record-Route and
      Contact) to build a Route header field.  The new INVITE to be sent
      (containing new digits) should not use any of these headers.  That
      is, the new INVITE does not contain neither To tag nor Route
      header field.  This way, this new INVITE can be routed dynamically
      by the network providing services.

But, again, what it you want to route to the same PSTN gateway?
 
later the RFC says:
3.5.  SIP to ISUP

   In this scenario (the call originates in the SIP network), the
   gateway receives multiple INVITEs that have the same Call-ID but have
   different Request-URIs.  Upon reception of the first INVITE, the
   gateway generates an IAM following the procedures described in RFC
   3398 [3].

   When a gateway receives a subsequent INVITE with the same Call-ID and
   From tag as the previous one, and an updated Request-URI, a SAM
   should be generated as opposed to a new IAM.  Upon reception of a
   subsequent INVITE, the INVITE received previously is answered with
   484 Address Incomplete.

 
For me, this is very problematic.  It prevents learning of the route
anyway.
So while one section says you shouldn't use the route, this "SIP to
ISUP"
section encourages behaviour that completely prevents learning of the
route.
 
 
 
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