BACKGROUNDERS | Updated: Tue, Feb 21, 2006 12:27 pm PDT


Last revised by Ali Farshchian, Feb 18, 06 at 7:16 pm PDT | Edit | Add New Term

There is a lot of debate about the two most popular types of VoIP; SIP and H.323, each of them has its own merits. Initially H.323 was the most popular protocol, though its popularity has decreased in the “local loop” due to its poor traversal of NAT and firewalls. For this reason as domestic VoIP services have been developed, SIP has been far more widely adopted. However in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being terminated over VoIP. So really SIP is a useful tool for the “local loop” and H.323 is like the “fiber backbone”. With the most recent changes introduced for H.323, however, it is now possible for H.323 devices to easily and consistently traverse NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously.

Where VoIP travels through multiple providers’ Soft Switches the concept of Full Media Proxy and signalling proxy are important. In H.323 the data is made up of 3 streams of data: 1) H.225.0 Call Signalling 2) H.245 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600ms) and packet loss will be high. However in signalling proxy mode where only the signalling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. These proxy concepts could lead the way to true global providers.

One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically to send a G.723.1 5.6kbps compressed audio path will require 18kbps of bandwidth based on standard sampling rates. The difference between the 5.6kbps and 18kbps is packet headers. There are a number of bandwidth optimisation techniques used such as silence suppression and header compression this can typically save 35% on bandwidth used. But the really interesting technology comes from VoIP off shoots such as TDMoIP which take advantage of the concept of bundling conversations that are heading to the same destination and wrapping them up inside the same packets. These can offer near toll quality audio in a 6-7kbps data stream.

This text is licensed under the GNU Documentation License. It uses material from the Wikipedia article “VoIP".

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