[SIPForum-discussion] Is voice path before 200 OK is valid ?

[Oscar Alonso Sánchez Hernández]

Hello

Please, refer RFC 3398 "7.1.6 Cause Present in ACM Message"


*7.1.6** Cause Present in ACM Message*

       SIP                       MGC/MG                       PSTN
        1|---------INVITE---------->|                          |
         |<----------100------------|                          |
         |                          |------------IAM---------->|2
         |                          |<=========Audio===========|
         |                          |<---ACM with cause code---|3
        4|<------183 with SDP-------|                          |
         |<=========Audio===========|                          |
                     ** Interwork timer expires **
        5|<----------4xx+-----------|                          |
         |                          |------------REL---------->|6
         |                          |<-----------RLC-----------|7
        8|-----------ACK----------->|                          |

   1.  When a SIP user wishes to begin a session with a PSTN user, the SIP
node issues an INVITE request.

   2.  Upon receipt of an INVITE request, the gateway maps it to an
IAM message and sends it to the ISUP network.

   3.  Since the ISUP node is unable to complete the call and wants
to generate the error tone/announcement itself, it sends an ACM with a cause
code.  The gateway starts an interwork timer.

   4.  Upon receipt of an ACM with cause (presence of the CAI parameter),
the gateway will generate a 183 message towards the SIP node; this contains
SDP to establish early media cut-through.

   5.  A final INVITE response, based on the cause code received in
the earlier ACM message, is generated and sent to the SIP node to terminate
the call.  See Section
7.2.4.1<http://tools.ietf.org/html/rfc3398#section-7.2.4.1>for the
table which contains the mapping from cause code to SIP response.

   6.  Upon expiration of the interwork timer, a REL is sent towards
the PSTN node to terminate the call.  Note that the SIP node can
also terminate the call by sending a CANCEL before the interwork
timer expires.  In this case, the signaling progresses as in Section 7.1.7.
++++



2008/6/11 Andrea Puddu <androjoker at hotmail.com>:

>  I understood the concept,
>
> but isn't the 183 mandatory to carry voice path before 200 OK?
> If the 183 is not mandatory, I can't understand its utility (compared to
> 180).
>
> Thanks,
>
> Andrea
>
>
>
>  ------------------------------
> Date: Tue, 10 Jun 2008 22:51:01 -0500
> From: oasanchezh at gmail.com
> To: shakthi_msc at yahoo.com
> CC: discussion at sipforum.org
> Subject: Re: [SIPForum-discussion] Is voice path before 200 OK is valid ?
>
>
> Hello
> Scenaries that involve RTP flow before 200 OK responses to an INVITE has
> several goals: puts announcements, per example for prepaid services, color
> ring back tone services. SDP carried in the provisional response (i. e. 180,
> 183) conveys the IP address of the media server that supplies the
> announcement. The 200 OK response to INVITE carries SDP with IP address of
> final destination.
>
>
>
> 2008/6/9 Shakthi <shakthi_msc at yahoo.com>:
>
>
> Is voice path before 200 OK is valid ?
>
> Pls look at the call flow:
>
>     A                                B
>
>               INVITE
>      --------------------------------->
>             180 Ringing
>     <---------------------------------
>            RTP (g.723)
>     <-------------------------------->
>             200 OK
>      <-------------------------------
>             RTP (g.723)
>      <------------------------------>
>             ACK
>       ------------------------------->
>             BYE
>       <-----------------------------
>             200 OK
>        ------------------------------->
>
> Is this a valid call flow?
> Can RTP come in to the scene before 200 OK?
>
> - Thanx
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>
>
> --
> Oscar Alonso Sánchez Hernández
>
>
> ------------------------------
> Windows Live Messenger Non frenare la tua voglia di comunicare, prova
> Messenger! <http://www.messenger.it/>
>



-- 
Oscar Alonso Sánchez Hernández
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20080612/2fba9b86/attachment-0002.html>