begin process at 2013 05 21 08:28:03
  Trouver un code source :
 
dans
 

RFC3398 :: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping

Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping

Voir toute la rfc dans une seule page

Page : 11 / 68

Télécharger le PDF

Auteur(s) : L. Ong, J. Peterson, G. Camarillo, A. B. Roach
Classé sous : Pstn, Signaling system no. 7, Ss7, Public switched telephone network
RFC 3398                  ISUP to SIP Mapping              December 2002


7. SIP to ISUP Mapping

7.1 SIP to ISUP Call flows

   The following call flows illustrate the order of messages in typical
   success and error cases when setting up a call initiated from the SIP
   network.  "100 Trying" acknowledgements to INVITE requests are not
   displayed below although they are required in many architectures.

   In these diagrams, all call signaling (SIP, ISUP) is going to and
   from the MGC; media handling (e.g., audio cut-through, trunk freeing)
   is being performed by the MG, under the control of the MGC.  For the
   purpose of simplicity, these are shown as a single node, labeled
   "MGC/MG."

7.1.1 En-bloc Call Setup (no auto-answer)

       SIP                       MGC/MG                       PSTN
        1|---------INVITE---------->|                          |
         |<----------100------------|                          |
         |                          |------------IAM---------->|2
         |                          |<=========Audio===========|
         |                          |<-----------ACM-----------|3
        4|<----------18x------------|                          |
         |<=========Audio===========|                          |
         |                          |<-----------CPG-----------|5
        6|<----------18x------------|                          |
         |                          |<-----------ANM-----------|7
         |                          |<=========Audio==========>|
        8|<----------200------------|                          |
         |<=========Audio==========>|                          |
        9|-----------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.  The remote ISUP node indicates that the address is sufficient to
       set up a call by sending back an ACM message.

   4.  The "called party status" code in the ACM message is mapped to a
       SIP provisional response (as described in Section 7.2.5 and
       Section 7.2.6) and returned to the SIP node.  This response may
       contain SDP to establish an early media stream (as shown in the
       diagram).  If no SDP is present, the audio will be established in
       both directions after step 8.



Camarillo, et. al.          Standards Track                    [Page 11]



Nos sponsors


Sondage...

CalendriCode

Photothèque

A découvrir



 
Développement réalisé par Nicolas SOREL (Nix) avec l'aide de : Cyril DURAND et Emmanuel (EBArtSoft), Merci à Vincent pour ses précieux conseils.
CodeS-SourceS.com© Toute reproduction même partielle est interdite sauf accord écrit du Webmaster
CodeS-SourceS.com© est une marque déposée tous droits réservés

Google Coop CodeS-SourceS Google Coop CodeS-SourceS
Temps d'éxécution de la page : 0,094 sec (4)

Nous contacter | Annoncer sur CodeS-SourceS | Mentions légales