[asterisk-dev] Why do you want to send SIP requests in the multimedia RTP stream????

[Olle E. Johansson]

22 jan 2013 kl. 20:58 skrev SVN commits to the Digium repositories <svn-commits at lists.digium.com>:

> Create common method for sending SIP requests in a session.
Why are you sending SIP requests in the multimedia path????

SIP is used to manage sessions - but the SIP status of a "call" is named "dialog".


RFC 3261:

Session: From the SDP specification: "A multimedia session is a
         set of multimedia senders and receivers and the data streams
         flowing from senders to receivers.  A multimedia conference is
         an example of a multimedia session." (RFC 2327 [1]) (A session
         as defined for SDP can comprise one or more RTP sessions.)  As
         defined, a callee can be invited several times, by different
         calls, to the same session.  If SDP is used, a session is
         defined by the concatenation of the SDP user name, session id,
         network type, address type, and address elements in the origin
         field.

Dialog: A dialog is a peer-to-peer SIP relationship between two
         UAs that persists for some time.  A dialog is established by
         SIP messages, such as a 2xx response to an INVITE request.  A
         dialog is identified by a call identifier, local tag, and a
         remote tag.  A dialog was formerly known as a call leg in  RFC 2543.


Using something closer to either RFC terminology or asterisk terminology - "channels"
makes it easier for new (and old grumpy) developers to understand your code and
refer back to the standards.

And it will make it easier for you as you discuss SIP stuff with other developers at SIPit 30 in North Carolina.

/O ;-)