The advent of Femto Cells

Home base station solutions, also called Femto cells have recently attracted a lot of attention. The idea is to give the mobile user a small box similar to wireless routers, which provides mobile cellular coverage at home. But why would mobile operators want to do that?

In order to answer this question, one has to look at what mobile operators are competing against. Fixed line carriers and DSL service providers are seriously threatening the mobile business by offering new products centered around Fixed Mobile Convergence, which in essence utilises wireless access technology to capture mobile users when at home, and divert their mobile originated voice and data calls to the fixed line network.

Mobile operators found themselves in a predicament: the traditional Macro cellular networks are not able to compete with what DSL service providers can offer, neither from a cost structure point of view, nor in terms of access speeds. (of course the mobile operators are partly to blame because they picked the wrong battle by trying to compete on the basis of access speeds rather than mobile centric applications).

A home base station or femto cell can help mobile operators in many ways. First they can compete directly with other home access solutions which rely on other types of access technology such as WiFi or WiMax. This is because low cost mobile handsets are widely available, whereas dual mode handsets with WiFi capabilities are pricey and still limited in variety.

The femto cell will also enable the operator to cater for subscribers where the service is most needed. Compare this with traditional macro deployment where the infrastructure is first built in places where subscribers are "likely" to appear and even before subscribers start using the network. A femto cell also circumvents the issue of outdoor-to-indoor propagation which typically kills the capacity of 3G and 3.5G systems.

The deployment costs associated with a femto cell are marginal in comparison with Macro cell deployment. This is because the customer is providing the "real estate" to keep the box, the power, as well as the backhaul (e.g. DSL subscription). I don't assume here that Femto cells will replace the macro layer, nevertheless the extent of investment in the macro layer will be considerably less.

Femto cells will also enable the operator to offer creative and disruptive tariffs to directly compete with fixed line. Home zone offers can also be more effective than traditional home zone offers using macro cells which extend for kilometers sometimes (because they rely on a cell ID parameter).

So where is the catch? I will discuss the downside in a future post.

Advanced Cellular Technologies Episode 4: Evolution of UMTS/HSDPA

3GPP is constantly thinking of ways to improve the performance of UMTS to compete with new emerging technologies that promise enhanced performance over legacy systems.

There are two evolution philosophies within 3GPP. One philosophy is start from scratch and design a new system with new architecture and air interface which do not have any legacy constraints. Another philosophy is to fine-tune existing architecture and introduce advanced features to the existing air interface to enhance its performance to the best level possible while maintaining backward compatibility. The driver for this second approach is to capitalize on existing infrastructure investment. While 3GPP is devising strategies to evolve UMTS on the long term to achieve a set of target performance goals, 3GPP is also thinking of ways to develop the existing UMTS standard to reach its full potential.

Near-term 3GPP enhancements
The near-term enhancements to UMTS are addressed in 3GPP Rel’6 and Rel’7. These primarily focus on the issues directly related to market deployment and the customer experience and to fine-tune and incrementally improve the performance of UMTS earlier standard releases to ensure that UMTS/HSDPA products and services are of high quality, perform correctly and are as fully featured as intended. Some of the goals of these enhancements are:

- Call set-up delay reduction for improved end-user experience.
- Signalling flow optimization.
- Support of real-time packet services such as VoIP.
- Increased system efficiency and overhead reduction.

Medium-term HSDPA evolution: HSPA+
HSPA+ is a manifestation of the evolved HSPA philosophy where the existing HSPA implementation is pushed to its limits by introducing various improvements to increase the efficiency of the system while maintaining backward compatibility. Some of these improvements are:
- Higher order modulation
- The use of advanced receiver such as Equalizers and IC.
- Possibly, the introduction of MIMO receivers and Receive diversity.

The main driver for HSPA+ is to capitalize on existing HSPA investment in infrastructure by focusing on backward compatibility and upgrade simplicity.

Long Term Evolution (LTE) of 3GPP
The main objective for LTE is to enhance the 3GPP standard to become a highly competitive packet-based radio access technology. 3GPP promises huge increases in performance and capacity with LTE. From a performance point of view, the main goals of LTE are:

Flexible spectrum usage with scalable system bandwidth from 1.25 MHz up to 20 MHz
Increased spectrum efficiency and peak data rates at cell edge. Target peak rates of 100 Mbps/DL and 50 Mbps/UL.
Reduced latency for both user and control plane: less than 10ms round trip delay for user plane between UE and the serving RAN node, less than 100ms transition time for control plane between inactive state and active state.

LTE philosophy is more “revolutionary” than HSPA+ in scope. In order to achieve the ambitious goals set for it, LTE takes a fresh look at system architecture and air interface access without the constraints of legacy systems. Therefore system architecture will noticeably change with a new radio access layer.

Although many aspects of LTE are still under discussion in 3GPP, some of the main attributes are:

• OFDM-based air interface (OFDM=Orthogonal Frequency Division Multiplexing).
• Flat IP system architecture. (Often this is described as SAE: System Architecture Evolution, and is a separate study item in 3GPP).
• Higher level modulation and state of the art receiver technology.

VoIP Episode 5: VoIP over UMTS mouth-to-ear delay

I have received some queries about the delay encountered by VoIP calls. Thanks for the interaction and questions, keep them coming.

In this post I will summarise the various reasons that cause delay of VoIP speech over a UMTS network, or over any network for that matter.
I should mention here that I am not trying to show that VoIP over UMTS is any better or worse than conventional UMTS (I will only do that for a high consultancy fee! ;-) ).

Let us first "dissect" a VoIP call:
Before you start a VoIP call (and during the call) signalling messages are exchanged between the two VoIP terminals and possibly with some other network elements. These are carried using the Session Initiation Protocol (SIP). When you talk into your handset speaker, your analogue voice signal is "encoded" as a string of bits. A number of bits are then packed together in one packet and all the necessary headers are added to it. These packets are then transmitted over the network. They propagate through one or more packet switched networks before they reach their destination. At the destination, the received packets are then reordered and a decoder transforms the digital bits to an audible audio signal.

The sources of delay are numerous:

• First, there is the coding delay, which is the time spent on transforming your speech into a digital stream of bits. This can be minimised by powerful DSP/electronics.
• Then there is the time it takes to compile a packet and add headers to it (packetisation delay). For example, if a packet contains X voice samples, then it has experienced a delay from the instant of the first sample to the instant the packet was complete and ready for transmission. Obviously the bigger the packets are, the longer is the packetisation delay.
• There is also the propagation delay. This is the time the packet takes to traverse the various networks before it reaches its destination. This could be the propagation time over the air interface, or over the various packet core elements. This is essentially determined by the speed of the different physical mediums over which VoIP packets are transmitted: wires, optical cables, radio ...etc
• In addition to this, there is the summation of queuing delays encountered by the packet at each node in the network(s). For example, we have seen in a previous post that a VoIP packet may have to queue for a HSDPA radio resource, or the packet may have to queue at some of the routers and switches before it reaches its destination. The various routers and switches along the route of the packet can be loaded differently, so some packets may be delayed more than others.
• When the packets arrive at the destination, they may not arrive in the right order, because they may traverse different routes before arriving at the destination and because some of them are delayed more than others. Therefore, the receiver has to buffer the packets in order to wait for the slower ones and reorder the packets before submitting them to the decoder. This is called "De-jittering". The de-jittering introduces another delay before the packets are decoded into an audible signal again.
• There is also the delay associated with call signalling. You can not start talking until it rings at the other side and the called party answers. Because of the large SIP messages this delay could also be long. The same issues arises when the call ends, which does not really affect the user perceptions as much as it keeps some network resources tied up for a while.

I hope this answered your questions. I will talk about how some of the advanced 3GPP features will address some of the delay issues in a coming post, so stay tuned!.

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VoIP Episode 4: VoIP over UMTS high speed channels

In the first episode we discussed how the various implementation options compare in terms of performance and complexity. We say that there are various necessary changes to the high speed channels to make them an attractive medium to carry VoIP traffic. Some of these changes are also necessary to reach VoIP capacities that exceed conventional circuit switched capacity. In this post I will discuss some of the suggested changes to the high speed channels to make them suitable to carry VoIP.

• Scheduler changes:
  3GPP defined the structures of the high speed shared channels such that they are optimal to carry large chunks of data. To make the system reactive to users' channel conditions, an intelligent scheduling entity that usually resides in the RNC is moved to base station. Thus if a user happens to be in good channel conditions, the scheduler allocates resources and send them to the user before the favorable channel conditions disappear. The problem with VoIP traffic is that the packets have to be delivered to the user pretty frequently in order to maintain the perception of a good quality connection. The packets can not be queued forever waiting until the user has good channel conditions. Therefore the scheduler algorithms, which were originally developed to be optimal to carry discontinuous chunks of data, have to be adapted to work well with VoIP traffic side-by-side. One of the concepts to enable this is called “priority queuing" which as the name implies, gives priority to delay intolerant traffic such as VoIP. Most of what I describe above applies to HSDPA (downlink). In the uplink, 3GPP included enough functionality in the HSUPA requirements to make it versatile and able to cope with VoIP traffic using a scheme called “Non-Scheduled Transmission”.
• HARQ Retransmission Reduction:
  The HSDPA channel uses also a concept called "HARQ retransmission", where lost packets are retransmitted (partially or completely). The idea is to optimize the air interface efficiency, so instead of sending few highly protected bytes that will be definitely received correctly, the transmitter sends a big chunk of bytes and hope for the best. If part of the data is received incorrectly, then the transmitter can retransmit the packet. This approach is one of the enablers of high speeds achieved on this type of shared channels (see note 1 below). It turns out that the HARQ retransmission concept is a challenge to smooth VoIP operation. This is because the VoIP connection is intolerant to delay, and obviously packet retransmission will increase the round-trip delay, which degrades the call quality. Packet retransmissions also increase the severity of packet jitter at the receiving end: one packet may arrive almost immediately and the next packet may be transmitted several times before successful reception happens. If you eliminate re-transmission altogether the channel efficiency drops and you end up with a reduced cell capacity. (Check note 1 below) .
• Reduce the possibility of code limitation (Fractional DPCH):
  Every user connected to the network using a shared channel requires a number of small dedicated channels to carry signaling and control information. These channels carry very little traffic, but they are numerous. If there are many users in a cell using the HSDPA channel, a large number of dedicated channel is required to support them, and it could happen that the system runs out of channelization codes (if you don’t understand the concept of channelization codes, check this). Therefore, 3GPP included a new feature called Fractional DPCH, which essentially multiplexes the control information of multiple users on one physical channel.
• Continuous Packet Connectivity (CPC):
  This is a huge 3GPP work item which includes many aspects that, although not necessary for VoIP to work, but will benefit its operation and increase the cell VoIP capacity. The most noteworthy of these aspects are:
  
  1. DPCCH gating: The control channels have a discontinuous pattern, which reduces the amount of interference and increases battery life.
  
  2. Modified HS-SCCH operation. HS-SCCH is a channel that carries HSDPA signaling in the downlink. This channel typically consumes large amount of power which eats up a part of the cell capacity. Therefore 3GPP came with various suggestions to reduce its effects by restricting it, or eliminate it altogether.
  
  3. CQI reporting reduction: CQI stands for Channel Quality Indicator, which is a message sent by the terminal to the bas station reporting the channel quality so that the base station can decide what to send to the terminal. As we discussed in the scheduler section above, successful VoIP operation is more concerned with the continuity of the traffic instead of mapping transmission to channel conditions, therefore reducing the rate of CQI signaling will reduce overheads without affecting VoIP quality. This means that more bandwidth is available for VoIP packets instead of overhead.
  
  All in all, CPC is expected to increase VoIP cell capacity by anything from 15% to 40%.
  

Note 1: There is an optimal packet loss rate that will cause the system to operate at the maximum possible capacity. On one hand, high packet loss rates mean that many packets are being retransmitted many times and hence the air interface resource is being wasted on these retransmissions. On the other hand, low packet loss rates mean that the transmitter is not taking chances and is protecting data more than necessary. This will lead to a lower cell capacity.

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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!.

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