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 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.

RTP

Real-time Transport Protocol (RTP) was originally designed for delivering multimedia content over the Internet. It is often used for streaming (delivering in real-time) audio and video content such as music and movies.

RTP always uses UDP (User Datagram Protocol) as the transport layer. It can be used in conjunction with both SIP and H.323 for delivering voice data in a consistent and reliable manner. It provides services for identifying the type of data, its sequence, and whether or not each packet has been delivered.

QOS

Quality of Service (QOS) in VoIP refers to the likelihood that voice data will be delivered quickly and up to a certain standard -- clear and without background noise. It is used for VoIP, multimedia streaming, and applications which require a high degree of reliability.

Essentially, QOS is provided by ensuring that enough bandwidth has been reserved for a particular application. There are two ways to do this -- providing more than enough bandwidth to meet all needs at all times, or reserve bandwidth when it is needed. The second option is more practical because there is no way to foresee exactly what network demands will be at any given time.

VoIP most often uses RSVP (Resource ReSerVation Protocol) to reserve bandwidth, although other solutions including VLAN (Virtual Local Area Network) and VPN (Virtual Private Network) are being used by some VoIP service providers.

RSVP

Resource ReSerVation Protocol (RSVP) is used to manage Quality of Service (QOS). RSVP is used to request a minimum bandwidth and latency from every Internet router between two endpoints. Those that comply will reserve resources for the data stream.

The Internet has mechanisms in place to monitor the signal path between any two points. When a reservation request is received, the routers along the path examine the state of the signal paths and decide whether they can accommodate it.  Once the reservation is accepted, the routers have to carry that data as specified. To do this they reserve the resources necessary to guarantee bandwidth. After receiving an RSVP the data path is monitored to make sure the data travels along the path as expected. If not, the reserve request will timeout after a certain period of time so that resources are not unnecessarily used up.

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