QoS issues in Voice over IP

COMP9333 Advance Computer Networks Mini Conference QoS issues in Voice over IP Student ID: 3058224 Student ID: 3043237 Student ID: 3036281 Student ID: 3025715

QoS issues in Voice over IP Abstract: This article presents an overview of the factors that negatively affect Quality of Service in the increasingly popular area of voice over IP networks. These factors are packet network related as well as terminal related including: delay, jitter, echo and packet loss. Keywords: QoS,VoIP,delay,jitter 1. Introduction Voice over IP (VoIP) is new technology that allows the passing of speech information over IP data networks. In contrast to common traditional telecommunications, which moves voice traffic over the special Public Switched Telephone Network (PSTN), VoIP technology means that the voice shares the same network with the data in a true voice data convergence. The Comdial IPFX system uses VoIP networks to interconnect one another. VoIP has all the well publicized advantages inherent in having a unified data network infrastructure to manage all communications needs. Care must be taken, however to ensure that the voice quality is preserved when a data network is used to pass voice information Traditionally, voice and data networks have been supported by two separated infrastructures. The voice transport over IP (VoIP) offers service providers the advantage of integrating their networks into a single infrastructure. The substitution of traditional voice networks by VoIP will depend on the capability to offer to the end user a similar quality. Figure 1. Networking IP FX Systems Using a Data Network The advantages of reduced cost and bandwidth savings of carrying voice over packet networks are associated with some quality of service (QoS) issues unique to packet networks [1]. The paper is organised as follows. The next section describes how delay impact the QoS in VoIP, then jitter, Lost Packet Compensation and echo in section 3,

4,5 respectively. Finally, section 5 concludes the paper. 2. Issue One: Delay The first factor which significantly affects VoIP quality is the delay that the VoIP data packets experience as they traverse the network. Delay is the amount of time that a data network packet takes to travel from the sender s application to reach the receiver s destination application [2].It is significantly factor affecting VoIP quality. Regular packet delays longer than 100 milliseconds begin to interfere with normal conversation [1]. Delay causes two problems: echo and talker overlap. Echo is caused by the signal reflections of the speaker s voice from the far end telephone equipment back into the speaker s ear. Talker overlap (or the problem of one talker stepping on the other talker s speech) becomes significant if the one way delay becomes greater than 250 milliseconds [1]. The end to end delay budget is therefore the major constraint and driving requirement for reducing delay through a packet network. Longer delays can cause echoes, which can make normal conversation difficult. The total end to end packet delay is the result of the many incremental delays along the connection path. Some small delay occurs during the digital signal processing of the speech signals by the IP FX. As the VoIP packets travel across the network, a number of sometimes large delays may occur. The following are sources of delay in an end to end, voice over packet call: 1. Network Delay This delay is caused by the physical medium and protocols used to transmit the voice data and by the buffers used to remove packet jitter on the receive side. Network delay is a function of the capacity of the links in the network and the processing that occurs as the packets transit the network. The jitter buffers add delay, which is used to remove the packet delay variation to which each packet is subjected as it transits the packet network. This delay can be a significant part of the overall delay, as packet delay variations can be as high as 70 to 100 milliseconds in some frame relay and IP networks [3]. 2. Accumulation Delay (Sometimes Called Algorithmic Delay) This delay is caused by the need to collect a frame of voice samples to be processed by the voice coder. It is related to the type of voice coder used and varies from a single sample time (.125 microseconds) to many milliseconds [3]. A representative list of standard voice coders and their frame times follows:

3. Processing Delay This kind of delay is caused by the actual process of encoding and collecting the encoded samples into a packet for transmission over the packet network. The encoding delay is a function of both the processor execution time and the type of algorithm used. Often, multiple voice coder frames will be collected in a single packet to reduce the packet network overhead. For example, three frames of G.729 code words, equaling 30 milliseconds of speech, may be collected and packed into a single packet [3]. Solutions: It s impossible to eliminate all delay from VoIP communications, since delay includes necessary processing time in the endpoints plus transmission time. Yet it is possible to reduce delay caused by network congestion and queuing mechanisms. Suppose, for instance, that intermediate nodes on the network service voice packets immediately, because the nodes recognize that voice packets require priority processing. Providing such special treatment for voice packets is one of the principles of Quality of Service (QoS) systems in VoIP communications. VoIP packet delay in the IP network can be quite large and can vary significantly from one network to another. Each router traversed in a packet s travel across a network adds an additional delay. The router s or switch s design and the configuration of its queueing (buffering) size will affect the delay each router adds to the voice packets. On the reception end, packet (or playout ) buffers can add more delay. These network delays can only be mitigated though careful network architecture, equipment selection and configuration. Minimizing the number of router hops along the path reduces delay. Small router queue sizes and high bandwidth connections also reduce delay. (Unfortunately, small router queue sizes may result in lost packets.) 3. Issue Two: Jitter Jitter is a variable inter packet delay timing caused by the network a packet traverses [3]. Jitter is caused by a variety of network factors, including congestion, lack of bandwidth, varying packet sizes, out of order packets, retransmission, packet loss and routing changes. Under such conditions, packets in the same flow may take different times to travel from a sender to a receiver. This variation in inter packet delay is called jitter. For the listener at the receiving end, jitter that s not mitigated by a jitter buffer results in the audio sounding choppy. Jitter buffers serve to alleviate audio problems caused by jitter by introducing a small delay buffer to provide some flexibility to the endpoint before it needs to assemble and play the audio. In an endpoint with a jitter buffer, excessive jitter may cause packet loss as a result of packets arriving too late, or may cause a small extra delay from dynamic adjustment for out of order or missing packets.

Figure 2, Example of Jitter This figure depicts packets arriving at an end point exhibiting varied delays in each packet transmission. Let us assume that the sender launches all packets into the network with 5 ms intervals. In the figure, packets A and B are separated by 50 ms, which means that there was 45 ms delay introduced by the network on packet B. Packet B and packet C were also launched 5 ms apart. However, they are received 100 ms apart. This means that the packet C suffered longer delay in the network, compared to packet B. Such variations in delay are called (packet) jitter. 4. Issue Three: Lost Packet Compensation Packet loss is a normal phenomenon on packet networks. Loss can be caused by many different reasons: overloaded links, excessive collisions on a LAN, and physical media errors, to name a few. Transport layers such as TCP account for loss and allow packet recovery under reasonable loss conditions. Network devices, like switches and routers, sometimes have to hold data packets in buffered queues when a link gets congested. If the link remains congested for too long, the buffered queues will overflow and data will be lost. The lost packets must be retransmitted and this adds to the total transmission time. Lost packets can be an even more severe problem, depending on the type of packet network that is being used. Because IP networks do not guarantee service, they will usually exhibit a much higher incidence of lost voice packets than ATM networks. In current IP networks all voice frames are treated like data. Under peak loads and congestion, voice frames will be dropped equally with data frames. Data frames are not time sensitive like voice frames and there is no point in retransmission of lost frames as in voice transmission, if a packet is late, it is as good as not reaching the receiver at all. If the lost packets are left untreated, the listener hears annoying pops and clicks. Some schemes called lost packet compensation schemes used to overcome the problem of lost packets are as under: 1. Interpolate for lost speech packets by replaying the last packet received during the interval when the last packet was supposed to be played out. This works well when the incidence of lost frames is infrequent. It does not work very well for bursty loss of packets.

2. Another way is to send redundant information at the expense of bandwidth utilization. The basic approach replicates and sends the nth packet of voice information along with the (n+1) the packet. This method has the advantage of being able to exactly correct for the lost packet. However, this approach uses more bandwidth and creates greater delay. 3. An alternative approach is to develop an algorithm in the digital signal processor that detects missing packets, and then replays the last successfully received packet at a decreased volume in order to fill the gaps. 4. Another problem is that of Out of Order Packets. When an out of order condition is detected in the network, the missing packet is replaced by its predecessor, as if it was lost. When the late packet finally arrives, it is discarded. 5. Issue Four: Echo in Voice over IP Another measurement that we scale the performance of voice over IP is echo. As you may have such experience that you can hear your own voice repeated when you make a phone call to your friend. It is an annoying echo. Let s examine echo before solve it. 1. Definition of the echo in VOIP An echo is the audible leak through of your own voice into your own receive (return) path [9]. Figure3. Simple Telephone Call with an Echo [9] As shown in figure 1, for each talker in a conversation there are two paths: transmit path Tx and receive path Rx. If Bob hears echo, which means some of his voice unexpectedly goes back before reaching Alice. 2 Analysis of echo in VOIP Echo in a telephone network is caused by signal reflections generated by the hybrid circuit that converts between a four wire circuit (a separate transmit and receive pair) and a two wire circuit (a single transmit and receive pair) [3].There are two basic characteristics of echo as follows: echo amplitude: Actually it is the energy of echo. The louder the echo, the more annoying is. round trip delay: It is the period of time required for speaker s voice to travel back to his/her earpiece. Here are some influences on voice quality brought by round trip delay. 0 25ms Masked by side tone and is unperceivable. 25 400ms Acceptable for users, some echo control devices required. > 400ms Unacceptable, must be avoided. [10]

3. Location of echo in VOIP As mentioned above, echo is a leaking signal that from Tx path to Rx path. Leak through only happened in analog circuits electronically and will never occur in digital circuits. Another factor to measure echo is round trip delay. If delay is less than 25ms, echo is not perceivable. In the topology of VOIP, like figure 2 shows, we can locate where echo happens. Figure 4. Echo in Voice over IP [2] As for the echo produced before voice inter into internet, round trip delay is such quick that can be neglect, we pay more attention to the tail circuits. 4. Solution to echo in VOIP There are a number of network elements that have effects on echo, such as loudness, hybrid transformers, telephones and routers. However, we have more interests on whether QoS can have some effects on echo. QoS on echo: As echo occurred in tail circuits not in internet, QOS can not eliminate echo because the packets for which QoS services are already include echo. However, QoS can reduce the round trip delay of echo by providing high level service to the voice. Remember the shorter the roundtrip time is, the less annoying is. Therefore QoS may improve the echo conditions by transferring echo from unacceptable to acceptable, or from perceivable to unperceivable. Echo Canceller: Echo canceller is used to eliminate the echo. It is running in voice gateway that can reduce or eliminate echoes leaked coming from tail circuits on the view of speakers. Therefore it can impede echo enter into internet. The echo canceller removes the echo by learning the electrical characteristics of the tail circuit and forming its own model of the tail circuit in its memory. Using this model, the echo canceller creates an "estimated echo" signal based on the current and past Rx signal (speaker s voice). [10] 6. Summary The benefits of converged networking are compelling, but as with most things of value, achieving these benefits requires careful preparation. Sending real time voice messages across a connectionless IP network requires the network to deliver information with an immediacy and predictability that data networks were not

originally designed to achieve. By using QoS tools, however, it s possible to achieve high voice quality over shared voice/data networks. To do so, network managers must condition the network to distinguish voice packets from data, and give the voice packets priority as they travel across the net. Managing the network for QoS enables companies to achieve in a connectionless environment some of the performance characteristics that make circuit switched networks effective for voice, while retaining the flexibility and economy of the shared multiservice network. These characteristics include minimal delay and the ability to prevent the entry of new data or voice class traffic from degrading calls already in progress.

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