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SIP Trunking Tutorial - SIP-T-SIP for Telephones: IP-to-PSTN via SS   Message List  
Reply | Forward Message #933 of 942 |
This presentation is part of the SIP-VoIP Essentials online course
available from NACSE at http://www.nacse.com or the version for
Channel Partners called NVBE or also onsite. Please email Tom Cross
cross@... for more details on all delivery and customizing
options.

This is an introduction to SIP Trunking and connections for
telephones and discussion of SIP telephony network with respect to
the PSTN-Public Switched Telephone Network via SS7-Signaling System
7. This tutorial will present the various SS7 protocols used in PSTN
signaling from a SIP phone. Later, other call processing scenarios
will be presented. The following are the steps in the SIP signaling
process to complete a call via the PSTN-Public Switched Telephone
Network. There are others but this is a start. First, the SIP
device sends an INVITE.

The GMC-Gateway Media Controller converts IP-Internet Protocol
packets to CAS-Channel Associated Signaling TDM-Time Division
Multiplexed channels signaling and media transmission with the PSTN.

A Q.931-SS7-ISUP-ISDN-Integrated Services Digital Network User Part
IAM-Initial Address Message is sent with the Routing label, CIC-
Circuit Identification Code and Message Type Code. Examples of
Message Type Codes include: Called Number, Calling Number, DPC-
Destination Point Code, OPC-Origination Point Code. As the IAM
passes through switching centers circuits (shown in green) are
reserved.

Once the IAM-Initial Address Message is processed at the DPC-
Destination Point Code (switch), an SS7-ISUP ACM-Address Complete
Message is sent to the OPC-Origination Point Code (switch). The ACM
is used also used to indicate the start of billing.

Upon off-hook connect, the reserved trunks are activated and an ANM-
ANswer Message completes the call setup/connect and communication
(conversation) begins.

RTP-Realtime Transport Protocol or SRTC-Secure RTP (encrypted media)
is the Session Layer protocol used for the voice transmission with
UDP-User Datagram Protocol and then IP for transmission.

During the call, various Call Progress messages such as CPG-Call
ProGress Message, COT-COTinuity Message, SUS-SUSpend Message, RES-
RESume Message, FOT-FOrward message Transfer, INR-INformation Request
message, INF-INFormation Message and others may be sent.

Upon RELease or disconnect (on-hook), a series of SS7-ISUP messages
are sent to teardown (release) the trunks, terminate billing, and
provide network management and testing for the next call.





Sun Aug 26, 2007 1:37 pm

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This presentation is part of the SIP-VoIP Essentials online course available from NACSE at http://www.nacse.com or the version for Channel Partners called NVBE...
talknettalk
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Aug 29, 2007
12:04 pm
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