15 Nov 2006

VoIP Episode 1: VoIP over UMTS

There has been talk in the market about introducing VoIP over UMTS networks. In this posting I investigate the viability of various VoIP over UMTS solutions. The three options presented here are: VoIP gateway, end-to-end VoIP over shared channels and end-to-end VoIP over dedicated channels.

Clarification of terms:
Before starting, let me clarify the term "VoIP over UMTS". I chose it because the prevalent term "VoIP over 3G" tends to be more generic and encompasses all technologies people consider as "3G" e.g. EV-DO. In this case I concentrate on UMTS only. Some people, when they refer to UMTS, they mean the 3GPP release 4 functionalities, and tend to refer to 3GPP R5 as HSDPA system and R6 as the HSUPA system. When I say UMTS here, I mean all the above. I should mention here that all the discussion that follows is primarily from a radio network point of view.


What drives service providers to think of the possibility of carrying VoIP over a UMTS radio network? Some of the reasons that I hear from time to time:

  • Operators think that VoIP can help them utilise their transmission network more efficiently.
  • They think that VoIP can help them squeeze a bit more out of their radio network.
  • They freak out when competing providers using other access technologies (WiMax, EV-DO ..etc) are offering VoIP. UMTS providers feel the urge to support VoIP to compete.
  • Convergence with their IMS networks.

Putting all the commercial drivers aside (competition, prices, availability ...etc), the four technical factors that influences the VoIP implementation are:

  1. Capacity: Can the system carry a reasonable and viable amount of VoIP calls?
  2. Complexity: How complex is the proposed solution?
  3. Call Quality: How does the VoIP call quality compare with users' experience?
  4. Seamless convergence: How easily can the implementation offer introduction of converged multimedia services and how readily is it compatible with existing and future network entities?

So what are the various ways to introduce VoIP/UMTS? There are three methods I can think of, which score differently in light of the four factors discussed above:

  1. Using a gateway solution:
    In this case, the VoIP traffic is terminated in the core network, and the voice is translated to circuit switched format to be carried over the radio network.
    UMTS was built from the onset to carry Circuit switched voice and packet switched data simultaneously, so by using the the UMTS circuit switched capability to carry voice traffic the call setup and release times are short. The call quality is also more predictable, sustainable, and is maintained at a reasonably high quality. The disadvantage is that the end-to-end VoIP capability is lost, and with it the ability to provide added services such as Contact Lists and Contact Presence Information. (It maybe still possible to offer the latter types of services on a separate bearer with an order of magnitude increase in network complexity.)
    Another advantage is that all the IP overheads are stripped off at the gateway, so there is less overhead to be carried over the radio, which is often considered the most expensive resource.
  2. Providing end-to-end VoIP over high speed channels:
    In this case, the VoIP call is carried all the way to the user terminal over high speed shared channels in the uplink and downlink (dubbed HSDPA and HSUPA in the 3GPP standard). The motivation is to use the channels originally envisaged for data applications to carry VoIP packets.
    HSDPA was originally conceived to increase air interface throughput by giving more channel resources to users who experience good channel conditions. Unfortunately, in VoIP case, this leads to degraded radio bandwidth utilisation, because users' channel condition are irrelevant. Instead, link continuity and sustained voice quality are more important. Therefore, new scheduling schemes have to be developed to suit VoIP type of traffic, which unfortunately defeats the whole purpose of a shared channel that is allocated to users based on their channel conditions. In a nutshell, the capacity performance of this option does not look very appealing. Another disadvantage is the increased call setup delay and packet round-trip delay. Essentially users' packets have to stand in a queue to get a portion of the shared channel resources. Luckily, this effect is less of an issue in the uplink (HSUPA) because of a standard feature called (Non-Scheduled transmission).

    Nevertheless, there are various proposed schemes to improve the shared channels in order to carry more VoIP calls. For example, Robust Header Compression (RoHC) reduces the amount of overhead in the IP packets. Another idea is carrying the call signaling over shared channel too, which reduces call delays. There are also other improvements to the uplink and downlink shared channels that will increase the capacity up by anything from 20% to 50%.

    The proponents of this option argue that the E2E attributes of the VoIP call are maintained. They also argue that on the long term the VoIP capacity is much enhanced by introduction of advanced receivers and diversity (transmit div, receive div or both).
  3. Providing end-to-end VoIP over dedicated channels:
    In this case, the VoIP traffic is carried over dedicated channels. This sounds a more plausible implementation than using high speed channels: The high speed channel model tries to blast high data rates to users in good conditions, which is great for data hungry applications, but is not advantageous to VoIP users who are more interested in quality than high data rate. The dedicated channels on the other hand adjust the power allocated to each channel (power control) , so each user gets just the right share of the radio resource (give or take). There is still the issue of packet overhead, therefore the capacity performance of this option is better than VoIP/high speed channels, but is still less than pure circuit switched voice.
    The E2E benefits are there too, so it is easy to introduce IM types of services.

So what is the verdict? VoIP over dedicated channels seems to offer the best trade-off between capacity and complexity. It also provides for reasonable VoIP call quality while maintaing enough attributes to enable easy multimedia convergence. On the long run, the introduction of enhanced features may cause the VoIP/HS implementation to perform better in terms of capacity and call quality.

I will discuss the various enhancements in a another post soon!.

© Copyright belongs to the author. Do not cite or quote without the express permission of the author

3 comments:

Anonymous said...

Would you explain me about the implementation of OSI layer ini 3G? Thanks.

virtual call center solutions said...

Thanks so much for this post. A combination of advice that I've heard before but always bears repeating; plus new tips that I really ought to consider

HGH Human Growth Hormone said...

Great article, I need to make a homework with this.