Two standards compete for IP telephony signaling. The older and more widely accepted standard is the ITU (International Telecommunication Union) recommendation H.323, which defines a multimedia communications system over packet-switched networks, including IP networks.The other standard, Session Initiation Protocol (SIP), comes from the IETF (Internet Engineering Task Force) working group. Another well established protocol by the name of MGCP (Also comes from the IETF) will be described in details along with the other mentioned protocols.
Voice over IP is a technology for the integration of voice and data in an IP-based network, e.g. LAN, and enables the exchange of voice in real-time in an Internet Protocol based network. Therewith it is possible to unify all means of communication - telephony and messaging - within one user interface in a convergent network. The IP-based network with e.g. LAN or WAN is already available. The VoIP-based PBX replaces or extends the conventional PBX and - like the conventional PBX - controls all the communication processes, e.g. the dial-up. The company does not need a PBX-cabling, spares the PBX-technician, expenses for costly feature-added telephones and for the productive operation. In the fields of Voice over IP, there are mainly two competing standards at the moment: H.323 and SIP.
H.323
H.323, which describes how multimedia communications occur between terminals, network equipment and services, is part of a larger group of ITU recommendations for multi-media interoperability called H.3x. The latest of these recommendations, H.248, is a recommendation to provide a single standard for the control of gateway devices in multi-media packet transmissions to allow calls to connect from a LAN to a Public Switched Telephone Network (PSTN), as well as to other standards-based terminals
The H.323 Architecture
A typical H.323 network is composed of a number of zones interconnected via a WAN. Each zone consists of a single H.323 gatekeeper (GK), a number of H.323 terminal endpoints (TEs), a number of H.323 gateways (GWs), and a number of multipoint control units (MCUs), interconnected via a LAN. A zone can span a number of LANs in different locations, or just a single LAN. The only requirement is that each zone contain exactly one GK, which acts as the administrator of the zone. The functionality of each component of the architecture is defined as follows:
Terminal: An H.323 TE is an endpoint in the network, which provides for real-time two-way communications with another H.323 terminal, GW, or MCU. This communication consists of control, indications, audio, moving color video pictures, and/or data between the two terminals. A terminal may set up a call to another terminal directly or with the help of a GK.
Gatekeeper: The GK is an H.323 entity in the network that provides address translation and controls access to the network for H.323 terminals, GWs, and MCUs. The GK may also provide other services to the terminals, GWs, and MCUs such as bandwidth management and locating GWs. The GK function is optional in H.323 systems.
Gateway: An H.323 GW is an endpoint in the network that provides real-time two-way communications between H.323 TEs on the packet-based network and
Multipoint control unit: The MCU is an endpoint in the network that provides the capability for three or more terminals and GWs to participate in a multipoint conference.
Signaling and Control
H.323 is an umbrella of the following four protocols:
● Registration Admission and Status (RAS): RAS is a transaction-oriented protocol between an H.323 endpoint (usually a TE or GW) and a GK. An endpoint can use RAS to discover a GK, register/unregister with a GK, requesting call admission and bandwidth allocation, and clearing a call. A GK can use RAS for inquiring on the status of an endpoint. There is also a mechanism for GKs to communicate with each other for address resolution across multiple zones. RAS is used only when a GK is present.
● Q.931: Q.931 is the signaling protocol for call setup and teardown between two H.323 TEs and is a variation of the Q.931 protocol defined for PSTN. H.323 adopted Q.931 so that interworking with PSTN/ISDN and related circuit-based multimedia conferencing standards such as H.320 and H.324 can be simplified. H.323 only uses a subset of the Q.931 messages in ISDN and a subset of the information elements (IEs). All the H.323-related parameters are encapsulated in the user-user IE (UUIE) of a Q.931 message.
● H.245: H.245 is used for connection control, allowing two endpoints to negotiate media processing capabilities such as audio/video codecs for each media channel between them. It is a common protocol for all H-series multimedia conferencing standards, including H.310, H.320, and H.324, and contains detailed descriptions of many media types. In the context of H.323, H.245 is used to exchange terminal capability, determine master-slave relationships of endpoints, and open and close logical channels between two endpoints.
● Real-Time Transmission Protocol: RTP is used as the transport protocol for packetized VoIP in H.323. It is adopted directly from IETF and is usually associated with Real-Time Control Protocol (RTCP).
Understanding H.323
The term signaling is used to describe the processes that initiate or terminate a communication session between two parties. If those parties are using the traditional telephone network, the signaling consists of on-hook, off-hook, ringing tones, busy tones, and so on, that communicate the status of one of the parties, or the status of the network.
H.323 endpoints—which could be Terminals, Gateways, Gatekeepers, or Multipoint Control Units—are packet-based, and as such, rely upon signaling messages, instead of signaling tones, to convey required control information to the desired destination. Note that control information is distinguished from data information, as the former is used to supervise and manage the connection between the two endpoints, and must be successfully completed before the data can be sent and received.
The portion of the network under the watchful eye of a Gatekeeper is called a zone, and large networks may have multiple Gatekeepers, and therefore multiple zones. When the Gatekeeper is not present, such as with an IP-phone-to-IP-phone connection, the signaling messages are passed directly between the two endpoints. When the Gatekeeper is present, the endpoints register with the Gatekeeper within their zone upon startup, using a process defined in H.225.0 called RAS, which stands for Registration, Admission and Status. For example, the Registration function occurs when the endpoint sends a Registration Request (RRQ) message to the Gatekeeper, and either a Registration Confirmation (RCF) or a Registration Reject (RRJ) message is returned. Other endpoint messages include the Admission Request (ARQ) asking for admission to the network, Bandwidth Change Request (BRQ), to request a specific amount of network bandwidth.
Once the endpoint has been registered with the Gatekeeper, an H.225.0 call signaling Setup message can be sent to the remote terminal, and H.225.0 Call Proceeding, Alerting, and Connect messages returned. At this point, a connection exists between the calling and called endpoints. Next, the endpoints must exchange their capabilities using the TerminalCapabilitySet [sic] signaling messages defined by the H.245 standard. A very broad set of parameters may be passed with these capability messages. For audio connections, the capabilities include the type of codec (G.711, G.728, G.729, etc.) plus parameters specific to that codec, such as the sampling rate, and the number of audio channels. For data connections, the capabilities include the data protocol in use (such as T.38 for fax or T.120 for whiteboarding applications), plus parameters specific to that data protocol, such as the transmission rate or data compression algorithm to be used. For video connections, the capabilities include the video codec type (such as H.261, H.262 or H.263), and plus parameters specific to that video codec, such as the number of samples per line, lines per frame, and so on. Similar procedures are defined which govern the call disconnection phase, but do not involve as many steps.
Interworking with the PSTN
Even though H.323 was designed for multipoint multimedia conferencing over packet networks, its usage has been primarily driven by VoIP applications, and interworking with the PSTN has been a focus from the very beginning. Unlike SIP, the GW to the PSTN has been an integral part of the H.323 architecture.
Interworking with PSTN usually concerns three call setup scenarios: H.323 TE to phone; phone to H.323 TE; and phone to phone via intermediate H.323 networks. In all cases, an H.323 GW is involved in connecting the PSTN with the Internet. Generally speaking, a GW needs to provide the following functionality.
- PSTN interfaces: This function includes the PSTN signaling interface that terminates signaling protocols such as ISDN Q.931, and the PSTN media interface that terminates media streams such as pulse code modulation (PCM) voice streams.
- VoIP interfaces: This function includes the VoIP signaling interface that terminates H.323 (including RAS, Q.931 and H.245), and the packet media interface that handles RTP.
- Signaling conversion: This function typically translates between ISDN Q.931 signaling and H.323 signaling for call control.
- Media transformation: This function typically translates between the 64 kb/s PCM streams and RTP streams of various speeds.
- Connection management: A major function implied by the above diagram is that a GW must internally coordinate between signaling flows and media transformations. This involves creating, modifying, and deleting the association between the PSTN and Internet flows during the lifetime of a call.
Functional Decomposition
Scalability: As discussed earlier, the bottleneck of scalability for H.323 GWs is media transformation. If we package the MGC and MG in separate boxes and use one MGC to control multiple MGs, we have in effect built a virtual H.323 GW that can handle more lines.
SS7 connectivity: This can be done by connecting the SG function to the SS7 network.
Availability: Decoupling the MGC from the MG increases availability in the sense that multiple MGCs can be used to control a single MG. If one MGC fails, but call states are kept in stable storage, one can apply traditional failover procedures to switch to another MGC. Graceful failover ensures that active calls in the MG s are not lost.
One-stage dialing: This is achieved through support for SS7 connectivity.
H.323 Protocol Suits
H.323 Protocol Suite
DVB - Digital Video Broadcasting for compatability with CATV
H.225 - call setup/termination
H.225 Annex G - address resolution between administrative domains
H.235 - security and authentication
H.245 - negotiates capabilities, codecs, flow control, and port numbers
H.261 - video stream for transport through RTP
Q931 - an ISDN protocol to manage setup/termination.
RAS - Manages Registration, Admission, and Status
RTCP - Real Time Control Protocol
RTP - Real-time Transport Protocol - carries the actual media over UDP
T.38 - Fax service - maps fax protocol T.30 to IP T.38 & visa versa
T.125 -
T.120 - realtime data conferencing - whiteboards, application viewing, application sharing.
H.450.1-12 - Supplemental services provided in H.323 are defined in the protocols H.450 .1 through .12 and include such functions as call transfer, hold, diversion and park, as well as call and message waiting.
Reference:
www.wikipedia.com (the free encyclopedia)
www.servonic.com
www.voipplanet.com
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