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Once channels are established, either a client or gatekeeper can request call services that is, a client or gatekeeper can initiate an increase or decrease of call bandwidth
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5 Terminate the call H245 call control and Q931 call signaling
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Call termination occurs as follows: Either party can terminate the call Assume client A terminates the call Client A completes the transmission of media and closes logical channels used to transmit media Client A transmits the End Session command (H245) Client B closes media logical channels and transmits the End Session command Client A closes the H245 control channel If the call signaling channel is still open, a Release Complete message (Q931) is sent between the clients to close this channel
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Figure 7-9 summarizes the steps of call setup
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Figure 7-9 Example of H323 call setup
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Both endpoints have previously registered with the gatekeeper Terminal A initiates the call to the gatekeeper (RAS messages are exchanged) The gatekeeper provides information for Terminal A to contact Terminal B Terminal A sends a Setup message to Terminal B Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission Terminal B sends an Alerting and Connect message Terminals B and A exchange H245 messages to determine master slave, terminal capabilities, and open logical channels
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Terminal A 1 ARQ 2 ACF
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Terminal B
3 SETUP 4 Call Proceeding 5 ARQ 6 ACF 7 Alerting 8 Connect H245 Messages RTP Media Path RAS messages Call Signaling Messages Note: This diagram only illustrates a simple point-to-point call setup where call signaling is not routed to the gatekeeper Refer to the H323 recommendation for more call setup scenarios
Signaling Issues for Voice over MPLS
Several new features are included in H323 version 2, as follows:
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H235 security and authentication, that is, passwords for registration with the gatekeeper H450x supplementary services such as call transfer and forwarding Fast call setup
Bypasses some Setup messages Triggered by the Q931 Fast Start message that contains basic capabilities
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Mechanism to specify alternative gatekeepers to endpoints The gatekeeper can request forwarding of Q931 information on direct routed calls Better integration of T120 (optional standard for data)
T120 channel is opened like any H323 channel
72 MGCP
The IETF RFC 2705 Media Gateway Control Protocol (MGCP) states that the MGCP is a protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents 6 MGCP is a master-slave protocol It assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent) It differs from SIP and H323, which are peer-to-peer protocols, but it does interoperate with H323 and SIP MGCP components include call agents (media gateways) and gateways, as depicted in Figure 7-10 MGCP is used between call agents and media gateways The call agent or media gateway controller (MGC) provides call signaling, control, and processing intelligence to the gateway It sends and receives commands to and from the gateway The gateway provides translations between circuit- and packet-switched networks The gateway sends notification to the call agent about endpoint events It also executes commands from the call agents A simplified call flow is as follows:1 1 When phone A goes off hook, gateway A sends a signal to the call agent 2 Gateway A generates a dial tone and collects the dialed digits 3 The digits are forwarded to the call agent
Figure 7-10 MGCP
Call Agent or Media Gateway Controller (MGC) SIP H323 Call Agent or Media Gateway Controller (MGC)
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MGCP
MGCP
Media Gateway (MG)
Media Gateway (MG)
4 The call agent determines how to route the call 5 The call agent sends commands to gateway B 6 Gateway B rings phone B 7 The call agent sends commands to both gateways to establish RTP/RTCP sessions At this stage, MGCP is a working document and not a standard The IETF and the ITU have decided to jointly mandate a single standard that is endorsed by both communities and known as MEGACO (IETF) and H248 (ITU) MEGACO is covered in more detail in Section 74
73 Session Initiation Protocol (SIP)
According to IETF RFC 2543, SIP is an application-layer signaling protocol that defines the initiation, modification, and termination of interactive, multimedia communication sessions between users7 SIP was designed for integration with existing IETF protocols, scalability and simplicity, mobility, and easy feature and service creation SIP is designed to be fast and simple in the (enterprise) core of the network SIP can support the following features and applications: basic call features (call waiting, call forwarding, call blocking, and so on), unified messaging, call forking, click to talk, instant messaging, and find me/follow me SIP is a peer-to-peer protocol (as are other Internet protocols) where a client can establish a session with another client By contrast, MEGACO is a master-slave protocol
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