identify call signaling and media stream flows
.Call signaling types
CAS(Channel Associated signaling)CSS -Common Channel signaling
CAS– signaling information is transmitted using the same bandwidth as the voice.
CSS- Signaling information is transmitted using , separate and dedicated signaling channel.
T1 CAS signaling steals bits from the voice channel to transfer signaling information
it is often called RBS (Robbed bit signaling ).
it is often called RBS (Robbed bit signaling ).
E1 lines have 32 channels , which break-down as follows :
à E1 DSO 1 – Used for E1 framing information.
à E1 DSO 2- 16 – Dedicated use for voice (no signalling)
à E1 DSO 17 – Used for voice signaling information.
The most popular signaling protocol used is Q.931 , which is the signaling protocol
used for ISDN circuits .
used for ISDN circuits .
CSS- Common channel signaling is used between voice Systesms worldwide because it offers more flexibility with signaling messages. it also allows PBX vendors to communicate proprietary messages , whereas CAS signaling Does NOT.
SIP
SIP is based on the request-response paradigm. The following sequence is a simple example of a call set-up procedure: 1. To initiate a session, the caller (or User Agent Client) sends a request with the SIP URL of the called party.
2. If the client knows the location of the other party it can send the request directly to their IP address; if not, the client can send it to a locally configured SIP network server.
3. The server will attempt to resolve the called user's location and send the request to them. There are many ways it can do this, such as searching the DNS or accessing databases. Alternatively, the server may be a redirect server that may return the called user location to the calling client for it to try directly. During the course of locating a user, one SIP network server can proxy or redirect the call to additional servers until it arrives at one that definitely knows the IP address where the called user can be found.
4. Once found, the request is sent to the user and then several options arise. In the simplest case, the user's telephony client receives the request, that is, the user's phone rings. If the user takes the call, the client responds to the invitation with the designated capabilities* of the client software and a connection is established. If the user declines the call, the session can be redirected to a voice mail server or to another user.
* "Designated capabilities" refers to the functions that the user wants to invoke. The client software might support videoconferencing, for example, but the user may only want to use audio conferencing. Regardless, the user can always add functions - such as videoconferencing, white-boarding, or a third user - by issuing another invite request to other users on the link.
SIP has two additional significant features. The first is a stateful SIP server's ability to split or "fork" an incoming call so that several extensions can be rung at once. The first extension to answer takes the call. This feature is handy if a user is working between two locations (a lab and an office, for example), or where someone is ringing both a boss and their secretary.
The second significant feature is SIP's unique ability to return different media types within a single session. For example, a customer could call a travel agent, view video clips of possible holiday destinations, complete an on-line booking form and order currency - all within the same communication session.
SIP Methods
The commands that SIP uses are called methods. SIP defines the following methods:
SIP Method | Description |
INVITE | Invites a user to a call |
ACK | Used to facilitate reliable message exchange for INVITEs |
BYE | Terminates a connection between users or declines a call |
CANCEL | Terminates a request, or search, for a user |
OPTIONS | Solicits information about a server's capabilities |
REGISTER | Registers a user's current location |
INFO | Used for mid-session signalling |
SIP responses
The following are SIP responses:
The following are SIP responses:
Inside a SIP message
The Request line and header field define the nature of the call in terms of services, addresses, and protocol features. The message body is independent of the SIP protocol and can contain anything.
SCCP :The Signalling Connection Control Part (SCCP) is a network layer protocol that provides extended routing, flow control, segmentation, connection-orientation, and error correction facilities in Signaling System 7 telecommunications networks. SCCP relies on the services of MTP for basic routing and error detection.
SCCP messages contain parameters which describe the type of addressing used, and how the message should be routed:
- Address Indicator
- Subsystem indicator: The address includes a Subsystem Number
- Point Code indicator: The address includes a Point Code
- Global title indicator
- No Global Title
- Global Title includes Translation Type (TT), Numbering Plan Indiciator (NPI) and Type of Number (TON)
- Global Title includes Translation Type only
- Routing indicator
- Route using Global Title only
- Route using Point Code/Subsystem number
- Address Indicator Coding
- Address Indicator coded as national (the Address Indicator is treated as international if not specified)
Protocol classes
SCCP provides 5 classes of protocol to its applications:- Class 0: Basic connectionless.
- Class 1: Sequenced connectionless.
- Class 2: Basic connection-oriented.
- Class 3: Flow control connection oriented.
- Class 4: Error recovery and flow control connection oriented.
The cisco VoIP structure :
- EndPoints (IP Phones, Wireless/cell Phones / Video Phone / IM Client )
- Applications (Voice Mail / Conference Call Apss / Call center Apps / 911 Services )
- Call Processing ( Unified Communications Manager /Unified Communication manager Express / UC500 )
- Infrastructure (ASA firewall / Voice Router /Gateway / Voice Switch)
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