QoS to be ensured in VOIP

Nowadays, people pay more and more attention to package-switch based telecommunication not only for its (theoretically) better performance than that of the circuit-switch-based telecommunication, but also for that it’s always FREE! The recent boom in wireless LAN deployment has generated a lot of interest in making VoIP available over such networks. Applications include:

Wireless phones in the enterprise; Dual mode cell phones that can connect to a WLAN for access to

higher bandwidth when available VoIP enabled personal digital assistants (PDAs).

Performance parameters are concerned when measuring Quality of Service (QoS). The current IP-based networks are not very suitable for transporting real-time traffic, such as voice. This is basically because of the connection-less nature of IP networks and the low QoS offered by IP networks. Because of the nature of wireless medium, the problem of VoIP is further accentuated. The performance parameters should be considered are: total end-to-end delay, delay variation at the receiver side (also referred as jitter), lost of damaged package and out-of-order package delivery.

Description of the Subject

This report introduces the following: An analysis of current VoIP performance improvement techniques under the wireless environment. Conclusions are drawn whether they can be adapted to wireless environment or improvement must be made. The following tasks are involved to achieve the above goal:

Analysis of the current VoIP improvement techniques to evaluate whether these mechanisms can be used under wireless environment.

The data collection path, which is the first step of network awareness. The algorithms to calculate the boundaries of delay and package loss.

The reconfiguration of the parameters of the session (mainly jitter buffer parameters) during the call/session on a per packet base.

A new linear regression algorithm with dynamic parameter to adapt to the fast changing wireless network.

An algorithm selector to select algorithm under different delay and package loss condition.

Analysis of VoIP QoS Improvement Techniques Quality of Service (QoS) is used to describe the performance of such system. QoS for VoIP includes jitter, latency, packet loss and bandwidth. It sets standards for optimizing network performance considering the above parameters. Delay is one of the important factors of QoS for VoIP networks, while package loss is also an essential factor under the wireless environment. According to nowadays broadband wireless network and future development, the bandwidth is less a problem. The QoS discussed in the previous chapter can be evaluated by evaluating certain performance parameters. As the tradeoff between time and area in computer hardware design, there is also a tradeoff between delay and package loss in VoIP system design.

Performance Parameters

In order to support real-time VoIP in the wireless IP environment, four main performance parameters should be considered:

End-to-end delay: The end-to-end delay encompasses: (i) the delay incurred in encoding (referred to as algorithmic delay), (ii) the delay incurred in packetization (function of the amount of speech data included in a packet), (iii) the delay incurred in the path from the sender to the receiver (propagation time, transmission time over network links, and queuing delays in network elements), (iv) the delay incurred in the playout buffer, and finally (v) the delay incurred in the decoder (usually negligible).

Lost/damage of packages: There are many reasons for this. The two major reasons are congestion (package is dropped because queue is full) and transmission error in the wireless interface (this is represented by BER). We

need to distinguish the two situations because they are caused by different reason and need different handling.

Delay jitter: Jitter is mainly interpreted as the difference in the total end-to-end delay of two consequent voice packages in the flow. This can be solved with buffers. This parameter is not handled in this report.

Out-of-order package delivery: The connectionless feature of the IP network causes out-of-order packages. This parameter is also not handled in this report. There are existing techniques, such as sequencing in RTP, that handle this problem.

Performance Tradeoff

The playout buffer removes the jitter using a jitter buffer mechanism. The jitter buffer removes the jitter in the arrival of the packets but does so at the cost of increase in the overall delay. There is a trade-off between the delay caused by the jitter buffer and the packet loss. A large jitter buffer causes increase in the delay and decreases the packet loss. A small jitter buffer decreases the delay but increases the packet loss. The size of the jitter buffer depends on the condition of the network, which varies with time. Typically the packet loss should be less than 10% for a good quality of speech.

Calculation of the Performance Parameters:

Delay was calculated using the capture data generated at the source node; whilst the remaining parameters were calculated from capture data generated at the destination. In order to demonstrate the method used, we will consider the following example capture files generated for the audio media at both workstations:

Delay: There is a practical difficulty in estimating the end-to-end delay of a transport channel, due to the lack of synchronization between the clocks at the channel endpoints. Instead, one can use the timestamps generated in the capture file of WinDump, and use the SR originating timestamp merely as a label to connect the pairs of SR and RR messages as illustrated in the following example:

If we observe the previous figure, we note that the SR message carries the originating RTCP timestamp of 4984.5664062500, and a WinDump timestamp of 08:13:50.831126. The corresponding RR message has an RTCP timestamp of 4984.5664062500+2.7451171875, indicating that this message was generated 2.7451171875 seconds after receipt of the SR message. At the source, the RR message has a WinDump timestamp of 08:13:54.212209. Our estimate of the effective round-trip time is then the sum of t1 and t2, which is seen to be 08:13:54.212209 – (08:13:50.831126+ 2.7451171875) = 0.635966 seconds.

Jitter: Jitter is measured by RTCP software and included in the RR messages sent by the receiver. As this value is measured in sampling units, in order to convert to time units, one must divide by the sampling rate of the media codec. Again using the example data, we have jitter values of 227 and 335 sampling units, which, when divided by ITU-T G.711 sampling rate of 8 kHz, correspond to 28.375 ms and 44.125 ms, respectively.

Packet Loss Rate: The packet loss rate is calculated from the data in two successive RR messages, which include both the highest packet sequence number received, and the cumulative total of lost packets.

Using the example data, in the interval between the two RR messages, the total number of transmitted RTP packets was (61413 - 61209) = 204, and of these the number of lost packets was (15 - 10) = 5, giving an instantaneous loss rate of 5/204 = 2.45%.

Special Issues Under Wireless Environment In the wired network VoIP research, the delay is the mean issue. Package loss and out-of-order package delivery sometimes is neglect for the following reasons:

The package loss caused by network in current broadband network is low. The package loss in the VoIP application layer is mostly caused by late-arrival.

The package lost caused by late arrival in the VoIP application layer can be solved together with delay by jitter buffer algorithm.

Studies of wired network loss characteristics have shown that the vast majority of losses in wired network are single losses, even when the loss percent is as high as around 10%. The situation is completely different in wireless environment. Because of the lossy nature of the wireless media, package loss is much greater in wireless network than in wired network. Furthermore, the retransmission mechanism of the MAC layer, IEEE802.11b for instance, converts the package loss into delay, which complicates the situation. Therefore, under wireless environment, if we see package loss and delay in the application layer, both of them must be used to represent the real situation of the network. In the wireless environment, the package loss must be considered separately and regarded as important as delay.

Approaches to Achieve QoS Improvement

There are many approaches to improve the performance of VoIP system. This section introduces these techniques and analyzes their performance in wireless environment.

Network Approaches Some of the standards of QoS are Resource Reservation Protocol (RSVP), DiffServ and 802.11e. RSVP and DiffServ are implanted in routers and some layer 3 switches i.e. network and upper layers. This however, does not help in wireless networks as QoS depends more on PHY and MAC layer than on layer 3.

A very efficient improvement of the network delay of real-time traffic is achieved by priority queuing of the network. This requires IP routers to support specific routing protocol. In an ad hoc wireless network, the routers are the terminal mobile nodes themselves. Because the intermediate mobile node cannot perform full function of a router, it is not practical to fully support priority queuing in ad hoc wireless network.

Frame Recover Approach

After a frame is lost due to certain reason, the receiver can have some approaches to recover the frame or reduce the damage caused by frame/package loss.

Forward error correction (FEC): In some cases, it has been suggested to even make use of forward-error-correction (FEC) codes on the transmitted packets to help protect against packet losses. FEC coding, however, does introduce some additional delays. Also, those packets that are transmitted over the “longer” paths will suffer larger delays than packets that happen to be sent over the shorter-delay paths. These features of the current FEC technology make it not suitable for wireless network, especially the infrastructure network with access points, which suffer heavy congestion when more users make VoIP calls through the AP. The following figure is from Nortel Networks.

Figure: Package Loss Effect on MOS value when applying FEC (Nortel Network)

Furthermore, the efficiency of FEC strongly depends on the rate and distribution of losses. In the wired network, the loss is mostly individual separated. Under such a situation, the FEC performs well when the total loss rate is under 2% (from Nortel Network). If the loss rate continues to increase, not only the performance of FEC decreases, the total delay also increases. In the wireless LAN environment, the package loss can be up to 20%. This indicates that FEC should not be used in wireless environment without careful planning. In order to have a ubiquitous performance improvement, the FEC is not used in this project.

Simple Error concealing: One of the simplest error concealing mechanisms is to regenerate the loss package by duplicating the previous package. This is a very simple but effective approach. The drawback of the error concealing is that this mechanism only works when loss rate is low and only single frame is lost. Though error concealing works under a very restrict condition, it is very effective without introducing any delay or adding complexity, because the mechanism is carried out at the receiver’s terminal. Therefore, simple error concealing can be used universally in wireless environment. However, when large amount of consequent packages are lost, the error concealing should be disabled. Instead, some mechanism should be used to inform the receiver that this period of the speech is corrupted.

Adaptive Codec Approach

The adaptive codec is specified in this report because this is the original idea of performance improvement on wireless environment. A lot of research and investigation is devoted to this field. An application-programming interface (API) for this mechanism is designed and implemented. The problems of applying the adaptive codec to wireless environment are:

The lack of appropriate codec for wireless environment. G.729 is default for VoIP but originally design for wired LAN. G.711 (PCMU) is the same as PSTN, but only appropriate for broadband network as it support data rate of 64kbps. There are other codec such as Adaptive PCM (ADPCM) which supports data rate range from 16kbps to 64kbps. But these codec are not well supported, because the current VoIP market is still VoIP on office LAN.

The hardware issues are very important. The current CPUs for wireless handset and PDA are not designed for real-time applications. Furthermore, the encoding and decoding are closely related to power consumption which is a nightmare for current wireless VoIP products.

Some basic compression codec and their performance parameters are shown in the following picture:

Table: Codec and MOS

This subjective quality measure is also referred to as Mean Opinion Score (MOS) and is given on a scale of 1 to 5, as defined in. The current PSTN rates around 4.0, which match the user’s satisfaction. The immediate conclusion is that the compression algorithms used on VoIP fit the basic requirement, not considering the packet losses and delay incurred with these losses.

The Linux operating system provides some very powerful tools to collect information of the wireless interface. Wireless Extension

Therefore, the complete picture of the system looks like:

Figure: System Architecture

In the above picture, the collection points might work together simultaneously in order to achieve most precise situation of the network. But, in the real programming level, the synchronization is a big problem. For the reason that the self-adaptive scheme should react quickly enough to the network, in order to achieve real-time network awareness, the collection points are ranked in priorities. We try to find a best solution to balance real-time factor and precision factor.

The challenges in deploying VoIP over WLAN stem mainly from issues related to access point congestion and to those that affect the link quality. The result is that WLANs experience significantly higher delay, with more network jitter and packet loss, than wired LANs. A focus is also on algorithmic delay. This source of delay is related to the speech codec used, and occurs because of framing for block processing, including look-ahead. Also, part of the algorithmic delay is the delay incurred by pre and post-processing; for example, echo cancellation, noise suppression and filtering.

Protocols Involved In VoIP

The VoIP system consists of many protocols. Among these protocols, the Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) are fundamental, while SIP is the most popular signaling protocol and RTP is the application layer protocol for the transport of real-time traffic.

Session Initiation Protocol

There are several competitive techniques for this purpose such as SIP, MGCP and H.323. Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify and terminate multimedia sessions or calls. These multimedia sessions include invitations to both unicast and multicast conferences and Internet telephony applications. SIP can be used in conjunction with other call setup and signaling protocols, and is described in RFC 3261. For the IP phony based on SIP, the following protocols are involved:

Signaling: SIP/SDP (SDP define the content/format of the signals) Media: RTP (IETF, also adopted by ITU-T) Transport: UDP, TCP is also supported

Here SIP is the signaling protocol very similar to HTTP, while SDP is comparable to HTML or XML. SIP is much more similar to HTTP rather than to legacy signaling both in terms of service model and protocol design.

Figure: Protocols Involved In VoIP

There are many advantages to choose SIP as the original protocol to enhance:

Compared with MGCP and H.323, SIP is much simpler, while SIP already supports most features of H.323, which is the most powerful one.

Its messages are very much similar to HTTP, which is also text-based transfer protocol. This feature makes SIP easy to design and easy to debug.

SIP has a feature to support mobility. Instead of mobile TCP/IP, which is part of the operating system, SIP can avoid modification of the OS by supporting mobility in the application layer.

As evidence, many dominating companies and organizations regard SIP as the future for VoIP. The most famous application based on SIP is Microsoft’s MSN messenger.

Because SIP is this popular, many dedicated organizations develop the protocol stack and the server as well as the user agency.

But SIP is previously designed for wired networks. In order to adapt SIP to the wireless LAN and even global wireless telecommunication network, improvement must be made.

Real-time Transport Protocol

RTP supports the transfer of real-time media (audio and video) over packet switched networks. It is used by both SIP and H.323. The transport protocol must allow the receiver to detect any losses in packets and also provide timing information so that the receiver can correctly compensate for delay jitter. The RTP header contains information that assists the receiver to reconstruct the media and also contains information specifying how the codec bit streams are broken up into packets. RTP does not reserve resources in the network but instead it provides information so that the receiver can recover in the presence of loss and jitter. The functions provided by RTP include:

Sequencing: The sequence number in the RTP packet is used for detecting lost packets

Payload Identification: In the Internet, it is often required to change the encoding of the media dynamically to adjust to changing bandwidth availability. To provide this functionality, a payload identifier is included in each RTP packet to describe the encoding of the media

Frame Indication: Video and audio are sent in logical units called frames. To indicate the beginning and end of the frame, a frame marker bit has been provided

Source Identification: In a multicast session, we have many participants. So an identifier is required to determine the originator of the frame. For this Synchronization Source (SSRC) identifier has been provided.

Intramedia Synchronization: To compensate for the different delay jitter for packets within the same stream, RTP provides timestamps which are needed by the play-out buffers.

Real Time Control Protocol (RTCP)

RTCP is a control protocol and works in conjunction with RTP. Click in a RTP session, participants periodically send RTCP packets to obtain useful information about QoS etc. The additional services that RTCP provides to the participants are:

QoS feedback: RTCP is used to report the quality of service. The information provided includes number of lost packets, Round Trip Time, jitter and this information is used by the sources to adjust their data rate.

Session Control: By the use of the BYE packet, RTCP allows participants to indicate that they are leaving a session

Identification: Information such as email address, name and phone number are included in the RTCP packets so that all the users can know the identities of the other users for that session.

Intermedia Synchronization: Even though video and audio are normally sent over different streams, we need to synchronize them at the receiver so that they play together.

Ways to improve your VoIP Quality of Service Before we get started, what exactly is Quality of Service (QoS)? According to Wikipedia it is: In the fields of packet-switched networks and computer networking, the traffic engineering term Quality of Service (QoS) refers to the probability of the telecommunication network meeting a given traffic contract, or in many cases is used informally to refer to the probability of a packet succeeding in passing between two points in the network within its desired latency period.As a VoIP call is basically a packet of data being transferred via the internet, your call is subject to potential problems such as packets loss, delays, jitters and errors. Therefore your connection to the internet and the devices used to connect to internet can play a part in reducing or improving your QoS.So in light of this, here a few ways that you can improve your VoIP QoS.

Upgrade Your Internet Connection

VoIP is highly dependent on upload and download speeds. If you have experienced hissing or delays during a call then your first stop should be with your current ISP. Check that they are not offering any free upgrades, in the UK many ISPs will actually upgrade your connection speeds for free simply to keep you as a customer. If you are unable to get a free upgrade, then you should either consider paying for more speed or moving to another ISP who offers greater connection speeds for the same price.

Improve Your Contention Ratio

OK, but what exactly is a contention ratio you may ask? You aren’t the only customer connecting to your ISPs servers. At any given time there

will be other customers connected to the same server. Some will be light users, sending a few emails and browsing the web. Others will be downloading music and movies, placing a heavy burden on that server. The more users there are connecting to a server at the same time, the slower your connection and the higher your contention ratio. To improve your call quality, look for a low contention ratio. Some ISPs will offer a 1:1 contention ratio, but you will probably have to pay handsomely for the privilege. My advice would be to ask for an upgrade to a contention ratio as low as possible, you’ll see big benefits! As long as it isn’t 100:1!

Optimize Your Computer for Broadband

A typical computer, fresh off the manufacturing line, is not optimized for broadband. Windows and Mac OS X are by default not programmed to handle the challenge posed by the larger packets and faster transfer rates of a broadband connection. However, there are many applications that you can download and install, which will change your OS’s networking settings to allow your computer to make the best use of your broadband connection. By optimizing your OS for broadband it will be capable of handling more traffic and thus improve your call quality. Here are few programs to try that will optimize your system:

Broadband Optimizer (OS X) SG TCP Optimizer (Windows) Web Optimizer (Windows) Upgrade Your Software

I know this sound so ordinary, but if you are a soft phone user then make sure you have the latest stable release. If you are using a beta version and have been experiencing issue with quality of service, my advice is to downgrade to the latest stable release. There are many benefits in keeping your software up to date as most new releases will have improvements in the codecs and compression technologies used for your calls. And if you are totally sick of what you are using, don’t forget that there is always an alternative.

Improve Your WiFi Signal

There have been a lot of WiFi phones announced recently that use a wireless network to make a call, typically with Skype. If you are using a WiFi phone for your calls then one of the best things you could do to

improve call quality is to improve your WiFi signal. There are numerous ways to achieve this; luckily we already have an article outlining 10 ways to improve your WiFi signal.

Use A DECT Phone

If you have tried to improve your wireless network so you can make better calls on your WiFi phone but have not had an improvement, perhaps you should consider a DECT VoIP phone. These phones do not need a wireless network; instead they work just like your existing cordless phone. The difference is that you can plug the base station into your computer via USB as well as plugging into the telephone socket. Another obvious advantage of using a DECT phone is that you can still make an emergency call with confidence as some VoIP services are currently unable to route an emergency call.

Get A New Headset This one is for those of us who use a soft phone and make calls while sitting in front of the computer. My basic premise is that headsets can be moody things, especially which refurbished model you got for a steal on eBay. Often the flimsy cable connecting the headset to the computer doesn’t last long. I find that the cable gets in the way all too often, meaning I have to tug it round the back of the desk or over a monitor. Just a word of advice here, light cables do not like getting run over repeatedly with your office chair. So if your call quality is poor, it could be time to bin your old headset and splash out on something new.

Drop WiFi, Use Ethernet

WiFi networks can be patchy, especially in areas where you are surrounded by concrete floors and walls. Problems with wireless networking are further exacerbated by interference from other electrical devices broadcasting a signal on the same frequency. If you have tried to improve your WiFi signal and failed, maybe it is time to try Ethernet. A wired solution reduces the possibility of interference significantly and can handle a lot of data intensive calls. Think about it.

Reduce Number of Simultaneous Calls

Every call you make places a demand on your connection. So it stands to reason that multiple simultaneous calls on the same network will add to the overhead. While reducing the number of calls is not an option in a business environment, you could try it at home.

Pause Any Downloads While On Call

Say you have been downloading a few torrents or movies off tunes, that will place a large demand on data transfer rates. Quite often I find that a call can drop out and experience delays while downloading large files, simply because my connection cannot handle so much data. Therefore, it would be an idea to pause any non-essential downloads while you make your calls then restart them.

Complain To Your VoIP Provider

This is pretty much the last chance saloon. If all else fails, get on the (PSTN) phone, call your VoIP provider and complain. Make it clear that you have exhausted all other methods of addressing the problem and have come to the conclusion that the fault is theirs and not yours. Ask them to get a technician on to the problem immediately; otherwise you will cancel your subscription. Personally I don’t like sounding like an arrogant, self-obsessed egomaniac on the phone, especially when the call handler is just doing their job. But at the end of the day, you are paying for an unsatisfactory service.