Protocols that work behind every internet phone call



Internet phone service is commonly used in many parts of the world. It uses internet to help people connect with each other globally. We all know that VoIP stands for voice over internet protocol. However, few of us really know what works behind the internet phones. All major VoIP services like Axvoice, Comcast, Vonage, and Skype use various types of these protocols. This article looks at all these protocols and the role they play in making the internet phone work.


It exclusively defines the protocols that are used for audio visual communication. There are many issues addressed by H.323 that include multimedia transport and control, call signaling and control, and bandwidth control. Many of the top voice and video conferencing equipment manufacturers also use H.323. This protocol’s flexibility makes it convenient to use in existing ISDN-based private branch exchanges. There are many control features offered by H.323 to audio and video phone calls. These controls include pick-up, hold, and call transfers. One of the great things about this protocol has been its early availability of particular set standards. These standards define dynamics of call model and the other additional services that business users of internet phone expect.

Media Gateway Control Protocol (MGCP):

This protocol has been specifically developed to work with public switched telephone network (PSTN). The convenience, MGCP offers is, that it utilizes the power of network in implementing PSTN over IP model which resides in call control center. Low intelligent endpoint devices are used to execute control commands. MGCP uses Real-time Transport Protocol for framing media streams. Similarly, for negotiating and specifying media streams that need to be transmitted through call session, it uses Session Description Protocol (SDP).

Session Initiation Protocol (SIP):

SIP or (Session Initiation Protocol) is used to control communication sessions for voice and video calls. The protocol can be effectively deployed to create, terminate, and even modify two or multiparty sessions. SIP supports single as well as multiple media streams. There are various ways in which SIP protocol facilitates communications like steaming multimedia distribution, instant messaging, video conferencing, and present information etc. SIP protocol is independent of the transport layer. It can run on different protocols like Stream Control Transmission Protocol (SCTP), User Datagram Protocol (UDP), and Transmission Control Protocol (TCP). SIP, being a text based protocol, inherits many elements from of the Simple Mail Transfer Protocol (SMTP) and Hypertext Transfer Protocol.
Real-time Transport Protocol (RTP):
The basic purpose of this protocol is to deliver video and audio content using the IP networks. Communication and entertainment systems that need media streaming use RTP protocol. The various systems that use real time transport protocol are telephony, television services, web based push to talk services, and video teleconferencing applications. RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP is used for media streaming, RTCP helps synchronization of multiple streams and improves the quality of service (QoS) by monitoring transmission statistics. Even port numbers are used for origination and reception of RTP. RTCP working with the RTP uses the next available odd port number.

Session Description Protocol (SDP)

Session Description Protocol describes the multimedia communication sessions. Unlike many other protocols, SDP is not involved in delivering the media; instead, it helps in negotiating between end points of format, media type, and all other related properties. These properties are known as session profile. SDP is designed to support the new formats and media types. In SDP, attributes are used for extending core protocol. Absolute time values are given in Network Time Protocol format.

Inter-Asterisk eXchange (IAX)

Inter-Asterisk eXchange protocol ensures establishment of VoIP connections between servers as well as client-server communications. IAX has been replaced with the IAX2. IAX 2 has the capability of carrying both signaling and media on the same port. One key advantage of using IAX2 is that it supports trunking. This trunking allows multiple calls to travel through single stream of packets which helps reduce IP overhead and overcome problems like latency. Internet phone service providers take many advantages of IAX2 like it helps them reduce the need to have large bandwidths for making internet telephony conversations.