Protocols Used by IP Telephony
At the moment, there is no clear standard specifying the operation of IP telephony protocols. Conventionally, we can divide IP telephony protocols into two groups: signaling and data transmission. We will try to consider the most common of them and used today from a practical point of view.
In addition, there are variations of this protocol for use in traditional SIP-T (Session Initiation Protocol for Telephones) networks (in RFC3372 and SIP-I (Session Initiation Protocol Internetworking). The main task of this protocol is the transparent transmission of IPS7 (ISUP) over IP Network.
SIP (Session Initiation Protocol)
Firstly, Session Initiation Protocol, the first version of SIP 1.0 was published in 1999 and was described in RFC 2543 recommendations by IETF. In 2002, the final recommendation of the SIP 2.0 protocol, described in the recommendation of IETF RFC 3261, was released. Since then, SIP has grown in many add-ons and extensions. SIP, being a client-server protocol, like HTTP and SMTP, works on the basis of consecutive requests-responses. Like HTTP, SIP is implemented using text tags – all SIP headers are transmitted as ASCII text, which simplifies its use in applications. At the moment, the SIP protocol has become fundamental in IP-telephony equipment, primarily for its conciseness and simplicity.
This protocol is used for communication of the elements of telecommunication networks: a gateway (Media Gateway) and a gateway controller (Media Gateway Controller). It supports various signaling systems for circuit-switched networks, including tone signaling, ISDN, ISUP, QSIG, and GSM. Hense, fixed as standard IMS protocol, along with SIP and Diameter. It is the heir of the MGCP protocol and it is used in the main networks of the IMS provider platforms.
Historically, the very first protocol for IP telephony, developed by the International Telecommunication Union (ITU) in 1996. In turn, H.323 covers the issues of voice, video data transmission via IP-networks. Therefore, professionals use this protocol less and less often, mainly for older analog PBX. The disadvantage of this protocol was its complexity and attachment to media data, in contrast to SIP.
Proprietary protocol for IP telephony used by Cisco in its telecommunications equipment. To some extent, third-party equipment Symbol Technologies, IPBlue, SocketIP, and Asterisk can work with this protocol.
IAX2 (Inter-Asterisk eXchange protocol)
The protocol developed for the operation of the Asterisk IP-PBX. Also, the peculiarity of this protocol is the adaptability to the translation of network addresses and the overcoming of NAT voice packets. Therefore, unlike SIP and H.323, specialists use only one UDP port 4569 for signaling and media data. As a result, specialists use this protocol networks with weak bandwidth and almost does not develop anymore.
Data transfer protocols:
SRTP (Secure Real-time Transport Protocol)
Secondly, Extension to the RTP protocol, providing encryption, authentication, integrity, and protection against repetition. Therefore, it is published as RFC 3711 and uses 5004port.
RTP (Real-time Transport Protocol)
A protocol intended for the transmission of audio and video streams over the Internet. RFC3550 describes it (before this in RFC 1889). Finally, the same standard describes the RTCP protocol (Real-time Control Protocol), which is responsible for QoS parameters between exchange participants.