Voip Protocols

Voice over IP uses internet protocol for transmission of voice packets across the internet. It can be achieved on any UDP/IP based network (LAN, WAN, Intranets etc). The communication between two points is digital communication. This requires voice coding and decoding at these two points. VoIP protocols are basically for call management and transport management.

VoIP signaling protocols are used to set up and terminate calls.They also carry information to locate the user and negotiate capabilities.

The VoIP protocol stacks derived from various standard bodies and vendors are

H.323
SIP
MEGACO
MGCP.

H.323

H.323 is the ITU-T standard, was originally for multimedia communications. Many extensions made it viable for Voice over IP. The series of recommendations defines protocols and procedures for multimedia transfer on the Internet
H.245 for control
H.225.0 for connection establishment
H.332 for large conferences
H.450.1 H.450.2 and H.450.3 for supplementary services
H.235 for security
H.246 for interoperability with circuit-switched services

H.323 is based heavily on the multimedia protocols (H.320 for ISDN, H.321 for BISDN, and H.324 for GSTN terminals) that preceded it. Multimedia data is exchanged via RTP. H.323 uses a binary representation for its messages, based on ASN.1 and PER. H.323’s complexity is because there is no clean separation of components of its services. Call forwarding requires H450, H225, H245. This complicates the process of firewall traversal. Its complexity is thus due to a duplication of its functions among the various associated protocols.

SIP- Session Initiation Protocol

The internet engineering Task Force developed a standard for Voice over the internet telephony. SIP solution voip; is compliant with IETF RFC 2543. This is an application layer protocol for creating, modifying and terminating sessions. In this protocol request are generated by the client and sent to the server. The server processes the request. The response sent to the client makes the transaction complete. Users in a SIP network are identified by unique SIP addresses. A SIP address is in the format of sip:userID@gateway.com. Users register with a registrar server using their assigned SIP addresses. The registrar server then provides the registration information to the location server upon request. SIP provides the following capabilities

Address resolution, name mapping and call redirection to determine the endpoint of the initiation.
Determines it the target endpoint is available. This is if the target endpoint did not respond to the call. A message is relayed back to the initiator as to why the target end did not respond.
SDP- Sessions Description Protocol determines whether the endpoint has media capabilities to enable a communication channel between the initiator and the end.
SIP supports mid call changes in media characteristic like change in code or enabling conferencing etc
It supports call transfer. It simply establishes a session to the new end point and terminates the session to the old end point and the new end point. The session therefore is not between the initiator and the new end point
Either and ATA, IP phone or PC can initiate SIP requests. Gateways that perform call setup and clearing on LAN networks and switched circuit networks can also initiate SIP requests. These are directed to a SIP server in the network. The redirection can be done through a proxy serve, redirect server or registrar server on the internet.
The enhancements that SIP provides are the ability to specify SIP or H.323 on a dial-peer basis, interoperability with unified call services (UCS), support for a variety of signaling protocols and interfaces.
The other support includes that of RTP/RTCP for media transport in VoIP networks. IP quality of service (QoS) and IP Security (IPSec) for SIP signaling messages
AAA- Authentication, authorization, and accounting and CDR-Call Data Record support.