VoIP Protocols

When shopping for VoIP services and equipment, you will often see references to H.323 and SIP. These are the two most common protocols used for handling VoIP calls, but there are also many others.

What Is a Protocol?

When we speak of protocols, we are referring to a set of rules that must be followed in order to allow two or more communication devices to 'talk' to each other. In the Internet and computer worlds, there are many different protocols which have been established.

The Internet has protocols for various purposes depending on the type of data that is being transmitted and its relative importance. Protocols can be layered -- used with each other to form a set of protocols which must be recognized at every point along the Internet pathways.

The basic protocol for the Internet is the Internet Protocol (IP). This allows computers to send data back and forth, but offers very little guarantee that the data will arrive intact. Other layers are used on top of IP in order to guarantee data integrity or speed of delivery. VoIP depends on rapid delivery of data packets, but is not overly concerned if a few of the packets are dropped en route. When data integrity is important (for example when transmitting program files) a protocol like TCP (Transmission Control Protocol) is used on top of IP. However, it is too slow for VoIP.

SIP

SIP stands for Session Initiation Protocol. It is becoming the standard for VoIP, and most VoIP service providers and soft phones use or at least offer this protocol.

SIP defines standards for a number of different services including caller identification, conference calls, call forwarding, and user mobility. SIP addresses are similar to IP (Internet Protocol) addresses and so can be used on web sites for 'Call Me Now' links.

As well as being able to handle voice, it is also suitable for transmitting multimedia such as video or music.

When used for VoIP, SIP assigns each user a unique address. This address is independent of actual physical location, so the same SIP address can be used by one user anywhere in the world. To initiate an SIP call, the caller sends an "invite" request to the person he wishes to speak to. The invite request is part of the SIP standard, and is handled transparently by the software or hardware that the caller is using.

As the other party is being searched for, response codes are sent to the call initiator. There are separate codes for searching, ringing, and success, as well as codes to indicate server failures or that the other party is not available.

Once the call has finished, a "Bye" command is issued to terminate the connection.

H.323

Like SIP, H.323 can be used for transmitting multimedia data. It was developed with multimedia data transmission in mind, something that makes it ideal for VoIP. It also has a number of features for interacting with PSTN (Public Switched Telephone Network). For example, included in its specifications is the ability to send and receive faxes -- something that poses technical difficulties with SIP.

It was originally developed for multimedia streams over a LAN, and was widely accepted in this role. The standards of H.323 have received wide acceptance and the specification continues to evolve. It is related to a suite of protocols which individually handle things like security, call signalling, and determining the capabilities of each party.

Even though H.323 was developed before SIP, it seems to be losing ground as a standard VoIP protocol. The main reason for this is the adoption of SIP by the 3rd Generation Partnership Project (3GPP) – the organization responsible for setting standards for 3rd generation mobile communication devices. In addition SIP is also much simpler than H.323.

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