VoIP Episode 3: UMTS imrpovements to carry VoIP

Looking at the viewing stats, the VoIP postings seem to have stirred some interest out there.
Hopefully you will find this posting equally interesting. In this one I will discuss some of improvements required for UMTS in order to carry VoIP traffic or to improve its performance.

• RoHC (Robust Header Compression) The various headers required to transmit VoIP packets typically add around 200-300% more bytes. As the name implies RoHC compresses the headers of the VoIP packets from tens of bytes to only a few. For a 12.2k codec the header overhead reduces to around only 30%. Two things to keep in mind though: first, the compression process is not instantaneous, it takes time for compression to be achieved, therefore a VoIP call may start with large uncompressed packets and then slowly converge to compressed packets. Think of the challenge to allocate network resources that vary in time (especially on the air interface). Second, for RoHC to operate correctly, feedback information has to be carried in the reverse direction, which means that the reduction in traffic gained in one direction is accompanied by a slight increase in traffic on the reverse direction. (This is especially the case when using RoHC R-mode, which is the mode advocated for wireless transmission) .

• Fast signalling: User experience is affected by multiple factors ranging from the quality of the call itself to the time the user has to wait to get his call through to the other party. VoIP calls particularly have large SIP signalling messages which does not only increased the demand on network resources, but also causes the call setup time to increase substantially. Therefore some changes to the way signalling is carried over the UMTS network are necessary in order to have a reasonable VoIP call setup delay. The most popular approach is to carry Signalling Radio Bearers (SRB) on high speed channels (HSDPA and HSUPA).

• Jitter management: Since IP relies on a connection-less type of transmission, packets take different times to propagate from one end to the other, depending on the route they take through the network, and on the instantaneous load of the various nodes the packets go through. This is not a problem for data applications like Internet browsing or email transmission, however it causes quality degradation in the case of VoIP. Therefore a mechanism is required to make the VoIP call more robust and tolerant to changes in packet propagation time (jitter). The simplest approach is to buffer the packets at the receiving end and re-order them if they arrive out of order. The downside is the increase in the perceived delay (which you may experience sometimes if you are making international calls using cheap calling cards - yes, they route VoIP on the internet!) .

In the next post I will discuss some of the changes needed to the high speed channels if VoIP is to be carried over these channels. Stay tuned.

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