This thread belongs on sip-implementors, to which it is redirected!

>1. How the billing is done in SIP, and how the billing information is coded
in the SIP messages.

There is no standard billing method.  There are a bunch of issues 
that are brought up constantly about billing.  Look back in the 
archive for several threads.  Generally, the most acceptable method 
is that one or more of the proxy servers emit billing records.  
Most UAs don't.  Billing is not carried in SIP messages, and thus not 
standardized, at least at the IETF.


>2. When a SIP User wants to make an International call, how does he select
different carriers. 
>    And where the carrier identification code is put.
SIP uses DNS to find the destination.  You don't use "carrier 
identification codes".  The service providers that provide the 
routing of the sip message determine how the messages arrive at 
the destination.  The media routes directly -- from one UA to the 
other.  Whatever service providers provide IP routing between the
UAs provide media stream services.  To "select" carriers, you select 
your IP service providers.  Note that SIP messages may traverse 
different paths than the media.

>3. What is the analogy of Termination ID of Megaco in SIP, and how to
represent? 
>    ( precisely I want to know about the  connection topology between Proxy
server and User Agent)
Megaco and SIP have pretty different models, and there is no 
real equivalent of Megaco's Termination ID.  However, you identify 
sources and destinations by URIs.  So, a URI is roughly the equivalent 
of a Termination ID.  This breaks down because you can have multiple 
SIP transactions in progress simultaneously between the same two 
URIs.  However, the standard defines a mechanism using a hash of 
multiple items that makes each transaction unique.  Then again, a proxy 
server need not be statefull, and thus "connection topology"
might be hard to discern.

Brian

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