Problem verification during execution of H.323 signaling

Problem verification during execution of H.323 signaling 1 ESAD KADUSIC & 2 NATASA ZIVIC & 3 NARCIS BEHLILOVIC & 4 ALIJA VEGARA 1,3,4 Faculty of Electrical Engineering in Sarajevo Zmaja od Bosne, 71 000 Sarajevo, BOSNIA AND HERZEGOVINA 2 Institute for Data Communications Systems University of Siegen Hölderlinstraße 3, Gebäude E, D-57076 Siegen GERMANY esad.kadusic@etf.unsa.ba, narcis.behlilovic@etf.unsa.ba, natasa.zivic@uni-siegen.de, alijavegara@hotmail.com Abstract: - Like every other network protocols and technologies, VoIP implementation must be kept within certain limits that communication services are on expected and acceptable level. For H.323 VoIP networks, there are two areas of performance which interest network engineers. Signaling performance affects length of time necessary for call establishment between two H.323 terminals. RTP performance affects delay and overall clarity of voice signal during VoIP conversation. This paper gives an overview of RTP performance and shows how to use H.323 decoded data in order to evaluate signaling performance. Key-Words: - H.323, H.225.0, H.245, Q.931, RAS, RTP performance, QoS, decoding, signaling performance 1 Introduction H.323 is one of the most important protocols in the VoIP world. It appeared under the ITU in 1996 and since has passed through many generation changes [1][2][3]. H.323 network consists of several elements. H.323 terminal is a basic mandatory network element. Besides terminal, standard defines also gatekeeper, gateway and MCU (Multi-point Control Unit) which are n principal optional network elements. The task of gatekeeper, if it exists, is to control other network elements (it can be compared with telephone switch). Gateway is a bridge between H.323 and some other network (for example PSTN), a MCU s task is to control conferences. Together with the advantages of new technologies protocols, come new challenges. For example, as a result of VoIP being relatively new technology, network engineers and administrators are fighting with multivendor competition, standard harmonization and interpretations, interwork problems, and all set of pther problems related to development and maintenance of VoIP networks [3][4][5][6]. Many of these problems are related to signaling. Signaling can be seen as a language, which control entities use for communication. The signaling process itself can cause normal process to stop, and therefore induce unacceptable delays in call establishment. Such inappropriate execution can also influence speech quality packet loss or excessive packet delay and jitter can cause poor sound quality, and codecs with low bit speed can additionally affect performance with increasing of packet loss [1]. Choice of codecs can influence on traffic load. For example, codecs which do not use compression will result in higher total traffic levels. Being unable to connect is simply a result of some error in very complex process of protocol exchange. Using of modern methods [1] for speech analysis and signaling protocols will make identification of problems which occur during making, keeping and terminating a call much easier as well as their solving. [1][8][9][10]. 2 H.323 Architecture H.323 is one of the ITU-T references which specifies complete architecture and methodology and also contains several other references. Together with H.323 is necessary to study some other references. Amongst the most important ones are H.225.0 and H.245 [2][3]. Overview of H.323 is shown on figure 1. Architecture includes H.323 terminals, gateways, gatekeepers, and MCUs. Fig. 1 The scope of H.323 and interoperability of H.323 terminals ISBN: 978-960-6766-64-0 146 ISSN: 1790-5117

Purpose of H.323 is to enable data flow between H.323 endpoints, where as H.323 endpoint is considered H.323 terminal, gateway or MCU. H.323 terminal is endpoint which offers communication in real time with other H.323 endpoints. Terminal is end user communication device which supports at least one audio codec and optionally can support other audio and/or video codecs. Gateway is H.323 endpoint which enables transfer services between H.323 network and other network types like ISDN or PSTN. One side of gateway supports H.323 signaling and works with packet data as determined by H.323 reference. Other gateway side represents a interface with other networks and supports transfer characteristics and signaling protocols of such networks. On H.323 side, gateway has H.323 terminal characteristics. On the switching network side it has characteristics of a node in such network. Translation of signaling protocol and data format from one to the other side is done inside gateway. This translation is transparent for all nodes in switching network and for H.323 network. Gateway also serves for communication between H.323 terminals which do not reside on the same network, and where communication has to be done over networks like PSTN. Gatekeeper is optional element in H.323 network. When it exists, gatekeeper control number of H.323 terminals, gateways and MCs (Multipoint Controller). Term control means that gatekeeper does the authorization of network access from one or more endpoints and can enable or disable some call from the endpoint under its control. It can offer services of bandwidth control, which brings to higher quality of service (QoS) if it is used together with bandwidth and/or resource management techniques. Gatekeeper also offers services of address translation, which enables usage of individual aliases inside the network. Set of terminals, gateways and MCs which controls one gatekeeper is called a zone. Figure 2 shows one such zone. Zone can be expanded on more networks or network segments and it is not necessary that all the elements inside a zone are physically connected. MC is H.323 endpoint which manages multipoint conferences between three or more terminals and/or gateways. For such conferences, MC enables media which elements share, by giving set of possibilities to different users. MC can change set of possibilities in case some other endpoints join the conference, or existing users leave the conference. MC does not make translation or function mixing. These functions are done by MP (Multipoint Processor) which controls the MC. MC can reside inside distinctive MCU or can reside inside the same platform as well as gateway, gatekeeper or H.323 terminal. For every MC there is at least one MP which works under the control of MC. MP is the one which process real data flow creating several output flows. MP is doing that through switching, mixing or combination of both. 3 Overview of H.323 signaling Figure 3 shows a model of H.323 protocol. It can be seen it consists of RTP, RTCP, TCP and UDP. From the picture it is clear that data exchange is done through the use of RTP over UDP, and off course where is RTP, there is also a RTCP. Fig. 3 H.323 protocol stack Fig. 2 An example of an H.323 zone On figure 3 we can also see other two protocols: H.225.0 and H.245. These two protocols define real messages which are exchanged between H.323 endpoints. These are generic protocols and as such can be used in different network architectures. When we talk about H.323 architecture, the way in which we use H.225.0 and H.245 protocols is defined by H.323 protocol. As mentioned before, real signaling protocols exchanged between H.323 elements are specified by ITU references H.225.0 and H.245. H.225.0 protocol consists of two parts. One part is a version of ITU-T reference Q.931, ISDN level 3 specification and is familiar to those who ISBN: 978-960-6766-64-0 147 ISSN: 1790-5117

know ISDN. This signaling is used for connection setup and termination between H.323 endpoints. This kind of signaling is known as call signaling or Q.931 signaling. Second part of H.225.0 is know as RAS (Registration, Admission and Status) signaling. This signaling is used between endpoints and gatekeeper and enables to gatekeeper to manage endpoints in its zone. For example, endpoint uses RAS signaling for registration at gatekeeper and gatekeeper uses RAS signaling to enable or disable endpoint access to the network resources. RAS signaling is always transferred over UDP, while call signaling can go through UDP or TCP. H.323 version 2, limits the use of call signaling on TCP. Setting up of TCP connection takes more time, which can bring to delay in call setup. To speed things up, in Annex E H.323 version 4 is specified a mechanism which checks up whether is UDP or TCP used for call signaling. In fact, both UDP and TCP can be used parallel. Sender sends its first message using UDP, and at the same time it sets up TCP connection. If, while TCP connection is established, there is no UDP answer, then the TCP connection is used. If the answer has come over UDP, then the original sender can assume that on the other end is supported H.323 version 4, and can continue to use UDP for call signaling. Sender then terminates TCP connection. H.245 is control protocol which is used between two or more endpoints. Main purpose of H.245 is management of data flow between H.323 users in a session. H.245 includes functions like ensuring that flow which is sent from one end to the other end is limited to the set of data which other element can receive and understand. H.245 works in a way that it sets up one or more logical channels between endpoints. These logical channels transfers data flow between users and have some characteristics like data type, transfer speed, etc. All three signaling protocols RAS, call signaling and H.245 can be used in setup, maintaining and termination of call. Different messages can be arranged. For example, lets look at the endpoint which wants to connect to other endpoint. First, endpoint can use RAS signaling to get authorization from gatekeeper. Then it can use call signaling to set up communication with other endpoint during call setup. And at the end, endpoint can use H.245 control signaling to exchange parameters with other endpoints to enable data flow. 4 Problem verification during execution of H.323 signaling Any advanced methods for solving of detected problems must be based also on detailed analysis of applied protocols in real time, which implies decoding of frames and packet on all protocol stack levels, statistic traffic analysis and especially implementation of expert systems in complex problem analysis. This chapter shows an example of two tests and usage of protocol analyzer, short instructions on a way how to solve testing problems of RTP and Quality of Service RTP performance, and gives an example of measuring signaling performance. We will use IP Telephony Analyzer for speech analysis and for signaling protocol analysis in IP networks. 4.1 Measuring of signaling performance In order to evaluate signaling performance we measure time necessary for connection setup. There are numerous conditions which can have opposite effect on time of call setup. They include: - Number of network nodes between H.323 endpoints. - Signaling load on gatekeeper or gateway. - Process performance of gatekeeper, gateway or terminal - Total network traffic and its influence pm packet delay and packet loss. We can use possibility of decoding to isolate problems in performance, connected to the abovementioned states, by measuring time necessary for network device to respond on message in which something is requested or to respond on other events in the network. Total call setup time is then sum of all individual measurements. To illustrate this, we will measure time necessary to complete signaling of routed call. The easiest way to measure is to use time stamps which are associated to ARQ and connection message when Advisor logs and decodes them. Figure 4 shows that the absolute time of ARQ message is (frame 173) 13:27:19.619409. Fig. 4 Absolute time of ARQ message On figure 5 we can see absolute connection message time (frame 162) 13:27:20.499236. Subtracting first time stamp from the second we can see that signaling ISBN: 978-960-6766-64-0 148 ISSN: 1790-5117

exchange of routed call between gateway 1 and gatekeeper lasted around 1 second. Fig. 5 Absolute connection message time Because we measured signaling process of routed call, this represents only roughly elapsed time from the moment when the destination address was called till the moment when the called side answered a call. Depending of user expectations, this can, but does not necessarily have to be considered unacceptable performance. Measurement also includes time which elapses between gateway/gatekeeper message exchange. Routed call signaling is only one part of total signaling process. Here, measured time does not include finding gatekeeper and terminal registration, initial communication and capabilities exchange, and can, but does not have to include delay times which are result of PSTN/PBX operations. 4.2 Overview of RTP Quality of Service RTP Statistics measurements show RTP QoS network performance. With this measurements, we can quickly see how interposing jitter and packet loss affect quality of service. Application show counters and statistics on users, sessions, and errors/alarms on the network. Also, it can be seen how much activities are performed in each of the categories. From these measurements we can quickly come to the decoded measurements from which can be seen detailed data of given frame. Charts in the lower window show interpose jitter and packet loss drawn by time and distribution. Each session tracks sent, received, lost and discarded packets as well as interposing jitter. Displayed details of the chosen user and session in the top of the working space show that packet loss has occurred. Warning window state on the excessive limit. Chart also shows interposed jitter of received packet, average for each sample period and packet loss in time. When the alarm or error occur, we can use directory tree to find out what caused the problem. 5 Conclusion and the future work Regarding the fact that the H.323 is relatively new area, network engineers and administrators are fighting with multivendor competition, standard compliances and interpretation, and a number of other problems related to development and maintenance of VoIP networks. H-323 is known on its rather complex signaling, high speed setup waiting and difficulties during implementation. Being unable to connect is simply a result of some error in very complex process of protocol exchange. H.323 defines several protocol exchange states between terminal, gateway and gatekeeper before setting up audio connection. Potential signaling problems during setup and connection phase, and information on their recognition should be searched in special signaling messages. This paper has given short overview of RTP performance and showed how to use decoded H.323 data in order to estimate signaling performance. Easiest way to measure delay time is to use time stamps which are associated to ARQ and connection message when the Advisor record and decodes them. Signalling of the routed call is only one part of entire signalling process. Any advanced methods for solving of detected problems must be based also on detailed analysis of applied protocols in real time, which implies decoding of frames and packet on all protocol stack levels, statistic traffic analysis and especially implementation of expert systems in complex problem analysis. Described techniques of problem solving by using the analyzer can very much ease the solving of practical problems which occur during setup, maintaining and ending of the call, as well as in speech analysis and signalling protocols analysis. Future work should consider problems of gatekeeper finding and registration, problems with call setup and forwarding, interworking problems, and in general VoIP performance and QoS. Mentioned future work guidelines open, without any doubt, very vast space for perfecting existing and creation of new methods and solutions in this field. Fig. 6 RTP Statistics measurements ISBN: 978-960-6766-64-0 149 ISSN: 1790-5117

References: [1] E. Kadusic, ''Identification and signaling problem solving in practical VoIP systems'', Master Thesis, ETF Sarajevo, 2006 [2] H.323 Standards, ITU (H.323v1-v6, H.225.0v1-v6, H.245) [3] D. Collins, ''Carrier Grade Voice over IP'', McGraw- Hill Networking, 2nd edition, 2003 [4] M. A. Miller, ''Voice over IP Technologies Building the Converged Network'', M&T Books, 2002 [5] D. E. Comer, ''Internetworking With TCP/IP, Principles, Protocols, and Architecture'', Fourt Edition, Prentice Hall, 2000 [6] J. Maeng and J. Ott, ''Interworking between SIP and H.323, MGCP, Megaco/H.248'' [7] Cisco Systems, ''Building a Voice and Video Signaling Strategy'' [8] Cisco Systems, ''Troubleshooting IP Telephony Networks for Cisco CallManager 3.0'' [9] S. Pracht, ''Troubleshooting H.323 Signaling'', Agilent Technologies [10] T1/E1 Trunk Analyzer & Monitor TAM 100, Teleprime, solution for troubleshooting ISBN: 978-960-6766-64-0 150 ISSN: 1790-5117