QoS in Voice over IP Zinal Patel Dept. of Electronics and Communication Engineering Institute of Technology, Nirma University Ahmedabad, Gujarat
[email protected]
Abstract— VoIP (Voice over Internet Protocol) service is a real time voice transmission in packets by utilizing telephone or PC by connecting telephone network and Internet. H.323 is the key protocol that allows interoperability of VoIP products. An overview of H.323 is presented in this paper. VoIP technology should be able to serve voice quality comparable to the traditional PSTN. This paper discusses the QoS parameters and the factors involved in making a high quality VoIP call. There is a tradeoff that must be made between delay and the efficient use of bandwidth. The quality of voice also depends on codec selection. Index Terms—VoIP, QoS, Internet telephony, H.323
I. INTRODUCTION Voice over Internet Protocol (VoIP) is related to the voice traffic in transmitted packets. This technology points out a huge gap from traditional circuit-switched transmission which is over public switched telephone network (PSTN) to the packet-switched transmission which is over Internet. The interest of the communication industry directed towards VoIP is stand over the two fundamental pillars of reason. The unification of voice and data traffic accepts the systematic use of data communication channels is the primary one. Secondly, major reason for Internet Protocol (IP) telephony is the global existence of protocol and its related protocols in user and network devices. On the other hand, the IP network follows a resource-shared mechanism while the traditional PSTN is resource-dedicated which make it cost efficient for each and every user of the network. There are various approaches to providing QoS (quality of service) in IP networks. Sometimes, the way to provide good IP network performance is through provisioning, rather than through complicated QoS protocols [1]. Suppose in the network, if link provocation is 30%, including all peak traffic conditions, in this case packets have to find path through it without any delays in the path. The design engineer must consider the number of channels which determine the capacity of the router components to deliver small voice packets ahead as well as the range of the frequencies of the inter-router links in determining the occupancy of the network. From the experiments we can say that if the occupancy is low, then the network behaves well as compare to the conventional one. The basic evolution of QoS features has maintained till now to achieve good performance of the network. In addition, the real-time traffic sometimes requires priority treatment for the network and makes it more efficient for the
each and every users of it. Bandwidth of the spectrum is also important parameter as its cost affect the performance of the network and as ahead the performance of each user. For some countries or states we can say that there are several cases where accessing of the links is expensive with the expensive broadband authentication in the network. Because of these problems in the network which degrades the performance of it as well as the users’, the QoS must advisable on the access links even if the core network or the server network is not much loaded. Wireless access links are especially expensive. On behalf of the performance of network we have the QoS for wireless mobile IP phone calls. For the practical implementation of the VoIP the replacement with standard PSTN telephony services, we must have to take care of the same quality of voice transmission should be there in the network is achieved. The difference in the quality of the service degrades the performance of the network and may be decrease the number of the users as they are not satisfied from the network service. This process is previously done with basic telephone services before the PSTN has established in the network. The QoS provides type of usage, attributes, and level of requirement as per user request in the network and so it is very important performance parameter on the basis of degradation and improvement of it inside the network. Several important number of factors are played important role in making a high-quality VoIP call with good performance of the network. These factors are; 1. 2. 3. 4. 5. 6.
Speech codec, Packetization, Packet loss, Delay, Delay variation, and The network architecture
Other factors such as the call setup signaling protocol, call admission control, security concerns, and the ability to traverse Network Address Translator (NAT) and firewall are important with above determined parameters to develop successful VoIP for the user in the network. The bandwidth and the jitter, basically known as the delay between the two adjacent packets in the network are very fragile to the VoIP application. User always demands the offers and the services which are based on the best performance of the network and also try to support it. They lack strict over the controlling of QoS in the network which degrades the performance of network. The packet loss, delay, and delay jitter are basically due to the congestion of the packets in the
networks which directly influence the quality of VoIP applications. The rest of the paper is organized such as in section II, an overview of VoIP architecture and H.323 signaling protocol is presented. Section III gives the description about quality of service in VoIP and QoS parameters. Lastly, section IV concludes the paper.
number of LANs in different locations, or just a single LAN. The only basic requirement is that each zone contains exactly one GK as shown in the figure, which placed at the job of administrative of the zone so that it improves the performance of the network.
II. VOIP ARCHITECTURE H.323 becomes very famous as it came out at the perfect time for the emerging of VoIP industry. It seems that the basic idea of the H.323 make the evolutionary growth in the VoIP and improves the performance of the network as well. The momentum of H.323 for VoIP was so great that by the end of 1996, most Personal Computer (PC) client software vendors were moving toward building H.323-compliant products to improve the performance. As same as the previous stage, interworking with PSTN was the focus from the very beginning since bypass of PSTN telephone charges was then regarded as one of the main economic drivers for VoIP. Current implementation of VoIP has two types of architectures, which are based on H.323 and SIP (Session Internet Protocol) frameworks, respectively. H.323, which was ratified by International Telegraph Union (ITU-T), is a set of protocols for voice, video, and data conferencing over packetbased network. Session Initiation Protocol (SIP), which is defined in RFC2543 of the Multiparty Multimedia Session Control (MMUSIC) working group of Internet Engineering Task Force, is an application-layer control signaling protocol for creating, modifying, and terminating sessions with one or more participants [3]. The basic architectures of the H.323 and SIP are the same on the platform of the implementation. They consist of three main logical components as the part of the implementation of network such as; Terminal, Signaling server and Gateway The difference among them is in specific definitions of voice coding which basically have the code of voice implementation, transport protocols for the determination of the services at the level of user satisfaction, control signaling has the control over the signal transmitted in to the channel contains data packets, gateway control, and call management of the network. The H.323 protocol is described as shown in the figure as described below. The main components of the H.323 network are; 1. The terminal, 2. The gatekeeper and The gateway and 3. The multipoint control unit. A typical H.323 network is designed with number of zones which are interconnected via a Wide Area Network (WAN). Here, each zone consists of a single H.323 gatekeeper (GK), a number of H.323 terminal endpoints (TEs), a number of H.323 gateways (GWs), and a number of multipoint control units (MCUs), interconnected via a LAN. A zone can span a
Figure 1: H.323 Networks with Local Area Network (LAN) [2]
Figure 2: Single H.323 Network [2]
H.323 is an umbrella of the following four protocols [2]: Registration Admission and Status (RAS): RAS is a transaction-oriented protocol between an H.323 endpoint (usually a TE or GW) and a GK. An endpoint can use RAS to discover a GK, register/unregister with a GK, requesting call admission and bandwidth allocation, and clearing a call. A GK can use RAS for inquiring on the status of an endpoint. Q.931: Q.931 is the signaling protocol for call setup and teardown between two H.323 TEs and is a variation of the Q.931 protocol defined for PSTN. H.323 adopted Q.931 so that interworking with PSTN/ISDN and related circuit-based multimedia conferencing standards such as H.320 and H.324 can be simplified. H.323 only uses a subset of the Q.931 messages in ISDN and a subset of the information elements (IEs). All the H.323- related parameters are encapsulated in the user-user IE (UUIE) of a Q.931 message. H.245: H.245 is used for connection control, allowing two endpoints to negotiate media processing capabilities such as audio/video codecs for each media channel between them. It is a common protocol for all H series multimedia conferencing standards, including H.310, H.320, and H.324, and contains detailed descriptions of many media types. In the context of H.323, H.245 is used to exchange terminal capability, determine master-slave relationships of endpoints, and open and close logical channels between two endpoints. Real-Time Transmission Protocol: RTP is used as the transport protocol for packetized VoIP in H.323. It is adopted directly from IETF and is usually associated with Real-Time Control Protocol (RTCP).
H.323 has some limitations. H.323 assumes that a gateway handles signaling conversion, call control, and media transcoding in one box, which poses scalability problems for large-scale deployment. H.323 also had no provision for SS7 connectivity, which hinders its seamless integration with PSTN.
kind of interfaces to QoS management, they do not provide functional QoS management mechanisms.
Figure 4: QoS management mechanisms of VoIP [3] Figure 3: The H.323 Protocol [2]
In order to provide carrier-grade VoIP services, the concept of a decomposed gateway was introduced where call control resides in one box, called the media gateway controller, and media transformation resides in another box called the media gateway. The Media Gateway Control Protocol (MGCP) was introduced in 1998 [2]. In June 2000, ITU-T SG16 and IETF defined the media gateway control standard, called H.248 or Megaco. III. QUALITY OF SERVICE The ultimate objective of VoIP is reliable, highquality voice service, as that of the PSTN. It is hard to achieve the same level of QoS as in PSTN. The main QoS issues are speech quality, service availability and usability. Voice requires lower delay, jitter and packet loss where as ordinary data transfer can be delayed without affecting the client’s requirement. Many factors determine voice quality, including the choice of codec, echo control, packet loss, delay, delay variation (jitter), and the design of the network. Packet loss causes voice clipping and skips. Some codec algorithms can correct for some lost voice packets. Typically, only a single packet can be lost during a short period for the codec correction algorithms to be effective. If the end-to-end delay becomes too long, the conversation begins to sound like two parties talking on a Citizens Band radio [1]. A buffer in the receiving device always compensates for jitter (delay variation). If the delay variation exceeds the size of the jitter buffer, there will be buffer overruns at the receiving end, with the same effect as packet loss anywhere else in the transmission path. For many years, the PSTN operated strictly with the ITU standard G.711. However, in a packet communications network, as well as in wireless mobile networks, other codec are also used. Telephones or gateways involved in setting up a call will be able to negotiate which codec to use from among a small working set of codecs that they support. QoS requirements of VoIP include packet loss, delay, and delay jitter. It should be pointed out; however, while the current H.323 and SIP frameworks support some
The quality of service parameters can be classified as connection quality and transmission quality parameters [4]. The connection quality parameters specify the quality related to call set up, call retain and call clear. These include dial tone delay, post dialing delay and rate of transmission. The transmission quality parameters show the quality and clarity after call set up. These are delay, jitter, packet loss, latency. Delay: Transmission time includes delay due to codec processing as well as propagation delay. ITU-T Recommendation G.114 recommends the following one-way transmission time limits for connections with adequately controlled echo (complying with G.131) [1]: • 0 to 150 ms: acceptable for most user applications; • 150 to 400 ms: acceptable for international connections; • 400 ms: unacceptable for general network planning purposes Jitter: Delay variation, sometimes called jitter, is also important. The receiving gateway or telephone must compensate for delay variation with a jitter buffer, which imposes a delay on early packets and passes late packets with less delay so that the decoded voice streams out of the receiver at a steady rate. Any packets that arrive later than the length of the jitter buffer are discarded. Since we want low packet loss, the jitter buffer delay is the maximum delay variation that we expect. This jitter buffer delay must be included in the total end-to-end delay that the listener experiences during a conversation using packet telephony. More than 60ms jitter is not tolerable. Packet loss: One of the serious problems on VoIP sending voice of internet is the loss of QoS and packets although the ultimate goal of VoIP is to provide quality of voice as in telephone; it is difficult to realize the quality due to the loss of packet. It is caused by the hybrid circuit where it changes from 4-wire to 2-wire [4]. Delayed echo is created from the other side of the telephone because of such circuit. Mean Opinion Score (MOS): It is one of the quality ratings on a scale of 1(bad) to 5(excellent). MOS is a better and reliable ways to estimate quality of service in VoIP networks. It takes
into account bandwidth, jitter, packet loss, codec selection at the same time.
system include the choice of codec and call signaling protocol. Packetized voice has larger end-to-end delays than a TDM system Since any packets that arrive later than the length of the jitter buffer are discarded, the jitter buffer delay must be set to the maximum delay variation that we expect, in order to achieve low packet loss probability. If end user wants to measure the overall quality, not only IP network to be measured but also the gateway, Gatekeeper, etc of VoIP network should be measured as well. REFERENCES
Table 1: Typical values of QoS parameters [3]
IV. CONCLUSION H.323 was the first VoIP standard in the communication industry. The H.323 architecture is still evolving in several areas such as the gateway decomposition architecture and integration of H.323 with PSTN. Other protocols such as SIP have also been introduced as alternatives to H.323. Providing reliable, high-quality voice communications over a network designed for data communications is a complex engineering challenge. Factors involved in designing a high-quality VoIP
[1] Bur Goode, "Voice over Internet Protocol (VoIP)", Proceedings of the IEEE, Vol. 90, No. 9, September 2002. [2] Hong Liu, and Patris Mouchtaris, "Voice over IP signaling: H.323 and beyond", IEEE Communications Magazine, 2000. [3] Xiuzhong Chen, Chunfeng Wang, Dong Xuan, Zhongcheng Li, Yinghua Min, and Wei Zhao, "Survey on QoS Management of VoIP", International Conference on Computer Networks and Mobile Computing, 2003. [4] Jeomgoo Kim, Inyong Lee, and Sichoon Noh, "VoIP QoS (quality of service) Design of measurement management process model", 2010..
