Voice over Internet Protocol (VoIP) is a method to carry voice traffic digitally over a network. Initial providers release services that emulate the traditional phone systems, changing out analog lines for digital.

More recently, call services like Skype have started offering new alternatives.Session Initiation Protocol is a signaling and control protocol for communications. It is most commonly used for voice and video delivery over an IP network but can also carry instant messaging.


SIP is primarily responsible for establishing and terminating these communications but not carrying actual voice traffic. SIP uses TCP and UDP port 5060 for plain text traffic and uses TCP and UDP port 5061 for TLS encrypted traffic. SIP packets employ a structure very similar to HTTP,including header fields and encoding rules. SIP resources are marked with a Uniform Resource Identifier, much like web resources. The URI scheme employed is usually in the form of sip:username:password@host:port.

SIP will often be employed on a digital office phone, also known as a user agent, to terminate to a local or remote SIP proxy server. A proxy acts as both a client and a server, taking a call request and forwarding it to the nearest endpoint it knows. Not only will a call control between a handsetand a proxy server utilize SIP, but often various gateways will also. A gateway is a method to connect to other mediums like outside providers or to the public switched telephone network. H.323 is also a signaling and control protocol for communication services.


It commonly uses TCP and UDP ports 1718 to 1720. H.323 was originally developed to facilitate teleconferencing but was quickly adjusted to be more encompassing. H.323 is most often used as a gateway for VoIP services. As the name implies, it acts as a gateway allowing a phone system to interface with the PSTN. Unlike H.323 gateways, MGCP gateways have very little intelligence, as they act more as a conduit for the phone system to control.

It utilizes TCP port 2428 and UDP port 2427. The Real-Time Transfer Protocol is what actually transfers voice and video traffic. RTP works in conjunction with the above signaling protocols as well as the RTP control protocol. RTCP periodically sends statistical information to participants in a media stream. It carries no voice traffic. It was originally developed in 1996 but was revised in 2003 as RFC 3550.


RTP employs an error concealment algorithm, which means it can sustain some packet loss without users noticing. Couple this with the need for audio packets to be transmitted as quickly as possible, it’s clear why RTP employs UDP as its delivery mechanism. RTP has a broad range of ports assigned:16384 to 32767, UDP. An RTP stream should be an even port number, and the next odd port should be the RTCP port.

The RTP profile type specifies the codecs used to encode the audio for the payload portion of the RTP packet. This can be methods such as G.711, G.723, G.726, and so on. There are a plethora of hosted SIP services out on the web now. For smaller organizations, nothing more than phones and an internet connection are necessary. For larger organizations, a single server can terminate SIP phones and also act as a voicemail server.

Larger enterprise generally utilizes multiple SIP and voicemail servers. Having VoIP experience is often a plus on a resume, so don’t hesitate to get your hands on some.

Hamza Arif
Follow us

Hamza Arif

Hey, i hamza arif student of telecommunication from BZU, i am good in Networking, Telecommunication and Web Development working on different projects and try my best to teach them to all of you.
Hamza Arif
Follow us

One thought to “VoIP: SIP, MGCP, RTP, H.323”

Leave a comment

Your email address will not be published. Required fields are marked *