Description Voice over IP (VOIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network. Using VOIP protocols, voice communications can be achieved on any IP network regardless whether it is Internet, Intranet or Local Area Networks (LAN). In a VOIP enabled network, the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. VOIP signaling protocols are used to set up and tear down calls, carry information required to locate users and negotiate capabilities. The key benefits of Internet telephony (Voice over IP) are the very low cost, the integration of data, voice and video on one network, the new services created on the converged network and simplified management of end user and terminals.
There are a few VOIP protocol stacks which are derived by various standard bodies and vendors, namely H.323, SIP, MEGACO and MGCP.
H.323 is the ITU-T's standard, which was originally developed for multimedia conferencing on LANs, but was later extended to cover Voice over IP. The standard encompasses both point to point communications and multipoint conferences. H.323 defines four logical components: Terminals, Gateways, Gatekeepers and Multipoint Control Units (MCUs). Terminals, gateways and MCUs are known as endpoints.
Session Initiation Protocol (SIP) is the IETF's standard for establishing VOIP connections. SIP is an application layer control protocol for creating, modifying and terminating sessions with one or more participants. The architecture of SIP is similar to that of HTTP (client-server protocol). Requests are generated by the client and sent to the server. The server processes the requests and then sends a response to the client. A request and the responses for that request make a transaction.
Media Gateway Control Protocol (MGCP), an IETF standard based on Cisco and Telcordia proposals, defines communication between call control elements (Call Agents or Media Gateway) and telephony gateways. MGCP is a control protocol, allowing a central coordinator to monitor events in IP phones and gateways and instruct them to send media to specific addresses. In the MGCP architecture, the call control intelligence is located outside the gateways and is handled by the call control elements (the Call Agent). Also, the call control elements (Call Agents) will synchronize with each other to send coherent commands to the gateways under their control. CableLab has adopted the MGCP for its PacketCable embbed clients in VOIP applications and the resulted protocol is called Network Based Signaling Protocol (NCS).
The Media Gateway Control Protocol (Megaco) is a result of joint efforts of the IETF and the ITU-T (ITU-T Recommendation H.248). Megaco/H.248 is a protocol for the control of elements in a physically decomposed multimedia gateway, which enables separation of call control from media conversion. Megaco/H.248 addresses the relationship between the Media Gateway (MG), which converts circuit-switched voice to packet-based traffic, and the Media Gateway Controller, which dictates the service logic of that traffic. Megaco/H.248 instructs an MG to connect streams coming from outside a packet or cell data network onto a packet or cell stream such as the Real-Time Transport Protocol (RTP). Megaco/H.248 is essentially quite similar to MGCP from an architectural standpoint and the controller-to-gateway relationship, but Megaco/H.248 supports a broader range of networks.
The SS7/C7 is the traditional signaling protocol for the circuit switched voice networks. To integrate the SS7/C7 network with the IP network, a group of protocols are defined, namely SIGTRAN (Signaling Transpor protocol). The key transport protocol in the SIGTRAN stack, the Stream Control Transmission Protocol (SCTP), has been applied in a much broader base after its creation.
In the past few years, the VOIP industry has been working on addressing the following key issues:
Quality of voice -- As IP was designed for carrying data, it does not provide real time guarantees but only provides best effort service. For voice communications over IP to become acceptable to users, the packet delay and getter needs to be less than a threshold value.
Interoperability -- In a public network environment, products from different vendors need to operate with each other for Voice over IP to become common among users.
Security -- Encryption (such as SSL) and tunneling (L2TP) technologies have been developed to protect VOIP signaling and bear traffic.
Integration with Public Switched Telephone Network(PSTN) -- While Internet telephony is being introduced, it will need to work in conjunction with PSTN in the foreseeable future. Gateway technologies are being developed to bridge the two networks.
Scalability -- VOIP systems need to be flexible enough to grow to the large user market for both private and public services. Many network management and user management technologies and products are being developed to address the issue.
Sponsor Source VOIP protocols are defined by IETF, ITU-T and some vendors.
Reference
http://www.cis.ohio-state.edu/~jain/refs/ref_voip.htm
Voice Over IP and IP Telephony References
Selasa, 14 Juli 2009
Voice over IP and VOIP Protocols
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