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VoIP Techniques and Challenges
Karama Said Mohamed
School of Engineering, Design and Technology
University of Bradford
other internet service at the basic level perspective. At the
transmitting end, the data (voice) undergoes compression,
V. What Is QoS?
QoS or Quality of Service is an overall set of network
standards and mechanisms that ensure that the servi...
Table 5: Bandwidth allocations for compressed
VoIP services operate using symmetrical bandwidth in
the uplink and down...
Figure 6: The acceptable range of delay
Figure 6 shows the acceptable delay for different
applications. Some of the delay ...
Type A: this type is classified as a constant jitter. The
packet to packet delay variation is almost constant.
Type B: thi...
be up to 400 milliseconds. This implies that VoIP is more
vulnerable to echo. [9][13]
When a portion of the talker’s voice...
IP Deployments available at:
[8] Quality of service, quality of Service for ...
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Research paper on VOIP Technology


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Research paper on VOIP Technology

  1. 1. VoIP Techniques and Challenges Karama Said Mohamed School of Engineering, Design and Technology University of Bradford Abstract Voice over Internet Protocol (VoIP) is a protocol aimed towards the optimization of voice transmission over the internet and other networks based on packet switching. The birth of VoIP came as an alternative for the much expensive Public Switched Telephone network (PSTN) for voice transmission. Putting into consideration the Quality of Service, (QoS) of the VoIP systems, factors like system capacity, jitter, and packet delay and loss and channel configurations are of utmost importance. These milestones together with security issues and channel bandwidth allocation and the question of reliability are great challenges facing VoIP systems. To master the VoIP systems, a clear understanding of the Internet Protocol (IP) is mandatory. Voice is carried by the RTP protocol added into the IP packet. In this research paper I will focus on the techniques employed in VoIP systems and the challenges that these systems face. With the voice transmission market still being in the transient stage from traditional PSTN to VoIP, new techniques are still being experimented and tested to improve quality of services offered by VoIP systems. The main point of focus is to come up with ways to deal with high traffic and the booming demand for the VoIP systems. Keywords: Voice over IP, PSTN, Quality of Service, Jitter, Packet Loss, Packet Delay, Internet Protocol, RTP, Channel Bandwidth I. Introduction The interest and need for Voice over Internet Protocol (VoIP) has been felt since the introduction of the first computer network. The main aim of VoIP Systems is to provide services that are either very hard or very expensive to implement using the traditional PSTN. The VoIP system is mainly based on offering voice communication by using the already existing internet Protocol (IP). [1] The internet has proved itself to be a cheap medium of sending data, like e-mails, globally over the years. Due to this fact, the VoIP system is invented to carry voice over this cheap media and to cut down the high costs associated with the traditional telephone lines. VoIP will enable people to communicate by voice all over the world at a much cheaper price than what people used to spend on normal telephone lines. This huge difference in cost has highlighted VoIP systems to be an interesting area for researchers. In contrary, cheaper costs of communication do not guarantee best services. Applications associated with VoIP have ever since its first usage, shown much poorer performance than conventional telephone services. Looking at aspects like Quality of Service, VoIP still needs a lot of improvements to stand up against the traditional telephone lines. The unsatisfying quality of data transmitted over the existing internet infrastructure could be due to the non-uniform nature of internet services that are available in different places in the world. [2] Currently, there is a lot of interest in the area of using VoIP over cellular networks. Due to the increase of users of VoIP through the internet and the ever existing need of cellular operators to maximize profits, introducing VoIP over cellular networks is a nut worth cracking. By introducing VoIP into cellular networks, existing cellular operators can easily and at a low cost switch to all-IP networks. This move would greatly reduce the operational costs and hence increase profitability. Not only the operators would benefit from this but also the users. The users will enjoy very much cheaper voice communication. This will be achieved by using devices such as the truphone. This device carries software that will enable to turn the mobile cellular phone into a VoIP phone while connected to the internet. [3][15] This all could be realized if the QoS of VoIP systems would be improved to the maximum. To achieve this desired QoS, things like system capacity, jitter, packet delay and loss and channel configurations must be put under serious monitoring. II. How Does VoIP Work? Due to the fact that VoIP makes use of the normal internet architecture, it functions more or less like any
  2. 2. other internet service at the basic level perspective. At the transmitting end, the data (voice) undergoes compression, analogue to digital conversion and then it is broken to data packets that carry distinct serial numbers. A lookup table will then aid the server at the transmitting end to determine the IP of the intended receiver. Once the server figures out the receiver IP it starts sending the data in the same manner as emails and other internet data. [14] At the receiver end, the data is collected and arranged as per the sequence numbers they carry and then it undergoes digital to analogue conversion to enable the receiver to hear the voice. Transmission using VoIP can be between two computers, two telephones or even between a telephone and a computer. With VoIP, a person can even use a PC to originate a call to a landline or vice versa. [Jain, 2004].figure 1 shows the overview of VoIP functionality network. Figure 1: basic VoIP functionality This is a basic view of how VoIP functions. In the next section I will elaborate in details the techniques used in VoIP systems. III.Carrying Voice over IP As I mentioned earlier, in VoIP systems, the voice is carried using the existing IP. This is made possible by adding the RTP header in the IP packet. Figure 2 shows the normal IP packet and the VoIP packet where the RTP header is included. Figure 2: RTP header added onto an IP packet The RTP is encapsulated in the UDP and IP.The voice bandwidth per conversation depends on CODEC and the sampling rate. [4] IV. VoIP Performance In order for VoIP to be able to stand up against the likes of UMTS, a very desirable QoS must be achieved. As mentioned earlier, one of the reasons for the poor performance of VoIP is that different regions in the world have distinct nature in terms of internet performance. To analyze in details the performance of this system we need to look into the following parameters;  System capacity/ Available bandwidth  Packet loss  Delay/Network Latency  Jitter  Echo  Security These mentioned factors are so far the major challenges faced by VoIP systems. All these parameters are variables that can be altered to achieve a desirable QoS. [14]
  3. 3. V. What Is QoS? QoS or Quality of Service is an overall set of network standards and mechanisms that ensure that the services offered are of high performance. Network administrators normally use the QoS mechanisms as a reference model to make optimum use of existing network resources to achieve the desired performance without the need for expanding or providing more resources to the network. Initially, quality in networks just meant equal treatment of the entire network traffic. This meant the network’s best effort was distributed equally to all traffic. This condition offered no guarantees for network performance characteristics such as delay and its variations, reliability and security. QoS is here to change the situation with the idea that different applications have different requirements and different users also have different needs. The main idea of QoS is that the effort of the network should not be distributed evenly to all the traffic in that some traffic needs to be given priorities over others. The main goal of QoS is to provide prioritized delivery services to the applications that require it. This is done by ensuring the provision of sufficient bandwidth, monitoring and controlling delay and jitter, and by minimizing data loss. In IP based networks, VoIP being one of them, there are two main models of QoS defined by the Internet Engineering Task Force (IETF), these are; the integrated Services (Intserv) and Differentiated Services (Diffserv). These two models have several mechanisms that ensure preferential services are given to specific traffic in the network. If VoIP applications achieve a desirable QoS then they will enjoy the following benefits;  Administrators will have a good control over the usage of network resources which will enable them to operate the network in a business perspective to maximize profits.  Applications and users who are time sensitive and critical will be provided with the resources they require at the same time other applications and users can have access to the network.  User experience will be improved as a result of improved system performance.  Due to the fact that existing resources will be put to optimum use, the general operating cost will be reduced. This will also ensure that there is a minimum need for expansions and upgrades. [5] VI. VoIP Challenges A. System Capacity/Available bandwidth. The provision of sufficient bandwidth for voice transmission is the first crucial step towards achieving a desirable QoS. The challenge here is that the available bandwidth is a limited resource. This implies that VoIP systems must be designed to make use of the available bandwidth without exceeding the limit, while at the same time carry real-time voice efficiently. Table 3 shows the bandwidth provisioning for VoIP. Table 3: Bandwidth provisioning for VoIP. To carry out a more accurate method for provisioning the layer 2 header is included into the bandwidth calculations. The results are as shown in the table 4. Table 4: Bandwidth provisioning for VoIP with the header 2 included in calculations.
  4. 4. Table 5: Bandwidth allocations for compressed RTP VoIP services operate using symmetrical bandwidth in the uplink and downlink. The main problem is that a bandwidth imbalance may exist in the uplink and the downlink even in the HSDPA phase. The system capacity also affects the provision of bandwidth traffic for each subscriber thus limiting the number of subscribers. In GSM and UMTS systems, the adaptive multi-rate (AMR) audio codec (12.2 Kbps) are extensively used in the CS voice services. In the case of VoIP protocol stack, the routing table protocol (RTP) and the user datagram protocol (RDP) are put in use. These two protocols are carried by the IP packet. Considering that the IP packet carries the RTP, UDP and the IP headers, the voice will require a bidirectional data rate of 32 Kbps or 64Kbps to carry out the transmission successfully. [6] B. Packet Loss In IP networks voice is treated as normal data. Due to this fact, the voice packets are vulnerable to the unfortunate cases of being dropped when the traffic is high and the network is congested. Re-transmission of lost data packets can solve the problem in data transmission, but this is not a solution for voice data. These solutions fail in voice data transmission because voice packets can contain a range of 40 to 80 ms of speech information. Packet loss greatly reduces the QoS of the systems. In systems like the ITU-TG.711 Vocoder, a standard for toll quality, a packet loss rate as low as 1% can cause a serious degradation in user experience. Other types of coders that carry out a more severe data compression tend to degrade more rigorously. [13] In the calculation of jitter, which I will discuss later in this paper, lost packets are usually neglected as they are considered to be packets with a delay magnitude of infinity and using them in the calculations will twist the calculations. Packet loss can be compensated in the end point by using algorithms like Packet Loss Concealment (PLC) or Packet Loss Recovery (PLR). Payload redundancy can be applied to counter packet loss but its use will require additional bandwidth. [7] In order to secure a sufficient bandwidth for the packets in a VoIP channel, a network device should be able to carry out identification of the VoIP packets. This implies that the VoIP packets should be able to be identified from all the other IP traffic. The network devices carry out this identification process by referring to the source and destination IP headers or the User Datagram Protocol headers. This process of packet identification is termed as classification and it is the basic foundation towards achieving a desired QoS. Another method of carrying out classification is by using the Resource Reservation Protocol (RSVP) mechanism. This mechanism carries out dynamic classification unlike the previously stated which is a static way of classification. After the classification process is completed by each hop in the network, each VoIP packet is then provided with the needed QoS. At this extent, special techniques can be assigned to achieve a priority queuing. Priority queuing ensures that any large data packets involved do not interfere with the ongoing voice transmission and minimizes bandwidth requirements in the fact that it compresses the 40-byte IP and UDP together with the RTP headers to 2 or 4 bytes only. [8] C. Delay/Network Latency Delay in networks is a condition that arises when voice packets take a longer time than expected to arrive at their destinations. This condition eventually results to distortions in the quality of voice. When transmitting voice packets, some of them get delayed and reach the destination later then expected. This delay may be caused by many factors and the main one being the underlying network. Delayed packets normally arrive at the destination late or never at all. QoS for voice transmission tends to be more tolerant on packet loss compared to text.)[9] The main known causes of delay are:  Codec  Queuing  Wait for packet being transmitted  Serialization  Jitter buffer
  5. 5. Figure 6: The acceptable range of delay Figure 6 shows the acceptable delay for different applications. Some of the delay causes can be dealt with but some of them there is no solution for them. Figure 7 shows the delay components at different levels of transmission. [10] Figure 7: delay components from source to destination. Delay can be classified into two categories: the fixed delay and the variable delay also known as jitter. Figure 8 shows the existence and causes of fixed delay in a network. The fixed delays are due to propagation, serialization and processing as shown in the figure. Figure 8: fixed delays in a network The propagation delay is normally about six microseconds per kilometre. Serialization delay occurs in the buffer to serial link. The processes that impose a delay are the likes of coding, compression, packetisation, decompression and decoding. The other type of delay is the variable delay, commonly known as jitter. Figure 9 shows the variable delay in a network. The main component here is the queuing delay which occurs throughout the network and it is greatly influenced by the packet size. The other thing that contributes to the variable delay are the de-jitter buffers that introduce variable delay so as to smooth out voice playout. [4] Fixed delays are out of our control but other delays can be reduced by the practice of marking voice packets as being delay sensitive. Another solution is to mitigate the effects caused by the jitter in a jitter buffer once they arrive at the destination. This process has a side effect of increasing delay. [1] Figure 9: variable delays D.Jitter A mentioned earlier, jitter is a variable delay caused mainly by queuing, contention and serialization along the network. In general terms, it is seen that jitter occurs most in links that are either slow or suffer from heavy congestion. QoS mechanisms such as queuing based on class, reservation of bandwidth and links that operate faster can greatly reduce the jitter problems in future. Until then jitter still remains a notorious drawback for VoIP channels. Jitter in real-time voice transmission can be classified into 3 types;
  6. 6. Type A: this type is classified as a constant jitter. The packet to packet delay variation is almost constant. Type B: this type is termed as the transient jitter. The main characteristic of this type of jitter is that it has a substantial incremental delay that may affect a single packet. Type C: this is the jitter composed of short term delay variations. Here the delay increases that affects numerous packets. Apart from this, a packet to packet delay variation may also be present. This type of jitter normally results from congestions and changes in routes. Transmit time jitter can occur in soft phones because the processes involved in the VoIP systems have to compete for the CPU time with other processes. This jitter is due to scheduling delays. Figure 10: experimental results of effect of packet congestion on delay (x = packet) Figure 10 shows how packet congestion in a network increases the delay time. The X’s in the graph represents packets and we see that where the packets are most congested, the delay is more. In the figure 11, another aspect is investigated that affects the delay, which is the access link congestion. Other main causes of delay are;  Sharing of load between many access links or IP service providers  Sharing of load within IP service  Inter-router load sharing  Routing table updates  Route flapping  Timing drift Figure 11: Access link congestion effects on delay The mostly used remedy for removing the effects of jitter is the use of jitter buffers. Jitter buffers are designed to erase the effects of jitter from the decoded voice stream. This process is done by buffering each individual packet for a short interval before it is heard by the receiver. As a result an additional delay is introduced and some packets get lost but jitter is solved. Adaptive jitter buffers are more preferable than fixed jitter buffers because they are capable of adjusting there size and to optimize the delay and discard tradeoffs. In terms of delay, both the fixed and adaptive jitter buffers are capable of carrying out automatic adjustments according to the changes in delay. For instance if a delay undergoes a step change of 19 milliseconds, then some packets may be discarded due to the change but the jitter buffer will be realigned fast. A jitter buffer is commonly looked at as a time window with the early side aligned with the recent minimum delay and the late side representing the maximum allowed delay before a packet is considered to be discarded [11] E.Echo Sometimes when users of VoIP make calls they could hear their own voice reflected to their phones’ speaker after a few milliseconds. This annoying phenomenon is known as echo. The time interval between speaking and hearing your own voice varies with the different causes of echo. A short interval echo does not cause so much harm but a longer one could completely destroy the conversation. A noticeable delay is the one which is loud and delayed. PSTN suffer from echo but not as much as VoIP systems. This is because PSTN has much lesser delay compared to VoIP. The maximum allowable delay for PSTN is about 10 milliseconds while that of VoIP can
  7. 7. be up to 400 milliseconds. This implies that VoIP is more vulnerable to echo. [9][13] When a portion of the talker’s voice is echoed back to him, this is known as talker echo. Listener echo happens when a portion of the talker’s voice is echoed back from the listener’s side and then proceeded by a second echo that causes a portion of the signal to reflect back to the listener. The end result s the listener hearing the talker’s voice twice i.e. echoed. The other type of echo is the convergence echo. This occurs at the beginning of a call and it occurs due to the delay in the echo canceller’s convergence. To solve the problems of echo, VoIP gateways make use of line echo cancellers to eradicate or minimize echo levels originating from analogue loops. Identifying a source of echo and checking its configurations are important processes involved in echo removal. [9] Echo cancellers normally face towards the PSTN tail circuit and they carry out elimination of echoes in the tail circuit on its respective side of the network. [8] F.Security Although it is much easier to secure a phone with VoIP than PSTN, a good number of consumer VoIP solutions do not support encryption yet. This makes it easier to carry out eavesdropping in VoIP and even change the contents of the data. Numerous open source solutions are available that facilitate the sniffing process of conversations through VoIP. A small degree of security is afforded by the use of scarce patented audio codecs that are not easily obtainable for open source applications. The use of this method of security is not proven effective. Compression is put in use by some vendors to counter eavesdropping. This method also only makes it difficult to eavesdrop but doesn’t prevent it. Encryption and cryptography are essential to ensure proper security in VoIP. There are possibilities to use the IPSec to secure VoIP by the use of opportunistic encryption. [12] VII. Conclusion VoIP is an evolutional step in voice communication that makes use of the widely spread and well establishes internet backbone. VoIP has managed to provide a much cheaper means of voice communication but still it is not wholly embraced by all. This might be because of its trade-off of low cost for poor QoS. The core reason for this low QoS in VoIP is that basically due to the fact that the internet was not designed for voice transmission. This is because the performance of VoIP is significantly hindered by factors like delay and packet loss. Delay has a much greater impact in the performance of VoIP due to the voice data sensitivity to delay. The nature of transmitting voice data over internet will always result to packet loss. The techniques used to counter the packet loss need to be closely monitored as most of them trade-off packet loss with delay. Apart from delay, jitter and packet loss, the question of security and reliability arises often due to the fact that the voice is transmitted over a public, widely spread media; the internet. To conclude, the PSTN system was designed for the sole purpose of carrying voice. Will the use of internet as a backbone to carry voice reach the standards of PSTN? Until the present, I can say VoIP can only be used in conjunction with the PSTN and not to replace it. VoIP may have a chance of replacing PSTN if and only if definite communication standards are set for VoIP, solutions for compatibility queries are defined and cross platform communication system is developed. References: [1]Voice over IP available at: ect=no [2] Term paper on voice over internet protocol available at: [3] [4] Voice over IP by David Lake, Cisco Systems Ltd available at: s/Heading/132 [5] [6] Leading Edge-VoIP over HSPA: running in the fast lane, By Li Xuanbo available at: 31&pid=61 [7] Overcoming Barriers to High-Quality Voice over
  8. 8. IP Deployments available at: [8] Quality of service, quality of Service for voice over IP available at: ons/QoSVoIP/QoSVoIP.html [9] [10] Quality of Service for Voice over IP (QoS for VoIP) Presented by: Dr. Peter J. Welcher. Available at [11] In depth: jitter. Available at [12] Examining Two Well-Known Attacks on VoIP By: Peter Thermos available at; acks_on_voip1/ [13] Olivier Hersent, Jean-Pierre Petit, David Gurle, Beyond VoIP Protocols: Understanding Voice Technology and Networking techniques for IP telephony, John Wiley and Sons, 2005. [14] Jonathan Davidson, James Peters Contributor: Brian Gracely, Voice over IP Fundamentals, Cisco press, 2000. [15] Ted Wallingford, Switching to VoIP, O’Reilly, 2005. .