ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/ ; Suresh.Leroy@alcatel.be +32/3/ ; Guy.Reyniers@alcatel.

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1 Contact: ALCATEL CRC Antwerpen Fr. Wellesplein 1 B-2018 Antwerpen +32/3/ ; Suresh.Leroy@alcatel.be +32/3/ ; Guy.Reyniers@alcatel.be Voice over (Vo) was developed at some universities to diminish the cost of long distance calls, for the price of decreased speech quality. Initially, it was considered as a gadget, but nowadays most network operators are taking Vo seriously and they are investigating and implementing solutions for integrating Vo and their classical PSTN. Even solutions where 2 POTS users are connected through Vo are possible. The Internet has changed the way of thinking in telephony networks and will play an even more important role in the future. It is clear that the internet technologies will also influence the architecture and implementation of future mobile networks. In this paper we investigate the possibilities for using Vo in the current mobile network (GSM, GPRS). The main problem for real-time applications in the Internet is the lack of QoS. In a mobile environment this becomes even more a problem, taking the limited bandwidth and error rates on radio interfaces into account. First, we give a short introduction on Voice over. Then we investigate 2 basic scenario s: using Vo end-to-end and using Vo internally in the mobile network. The basic conclusions of this investigation of the use of Vo in today s GSM and GPRS network are also applicable to the future UMTS network. Voice over allows users to speak to each other using the Internet (or an intranet) as the transport network. It was invented for personal computer and workstations, (semi-)permanently connected to an internet, to enable people to communicate verbally and simultaneously allow then to use their data application. Recently, telecommunication operators have seen the opportunities (and the threats) Vo brings and they started the implementation of Vo gateways in order to allow communication between a PC and any telephone connected to a PSTN. Some operators even offer a transcending to Vo for normal voice calls at a reduced cost (and sacrificing some quality). Recently the scope of Vo is extended to multimedia

2 services. There are two mayor frameworks for supporting Vo; one driven by ITU (H.323) and the other driven by the IETF. H.323 is designed to support multimedia applications over packet networks with no QoS (main focus on ) and is more telecom oriented than the IETF model. The main problems with H.323 are the scalability, complexity and interoperability between different implementations. The H.323 protocol was originally designed to be used in a LAN environment not a WAN. The reasons for the bad scalability of H.323 are threefold: - One gatekeeper controls all the terminals and gateways within one zone - Every gateway has to implement the complete H.323 protocol stack independent of the size of the network. - Because there is no signalling gateway, the signalling and data traffics are terminated in the same device. The importance of H.323 is diminishing with the development of a new protocol: MGCP (Media Gateway Control Protocol). MGCP solves the scalability problem by putting the control of gateways into an external device (the call control agent) thereby moving the intelligence outside the gateways. The use of Vo has some important advantages for the operators. Namely, the management of only one network for data and voice and the lower cost of IT equipment compared to telecommunication equipment. Using packet switched (PS) networks for the transport of voice (real-time data) is also more efficient in terms of bandwidth usage than circuit switched (CS) networks. Bandwidth is consumed by a user when needed, not pre-allocated. This, of course, implies that quality can not be guaranteed unless special techniques are used. Vo also allows the users to select a codec different from a PCM codec. This allows users with powerful machines to select a more performant codec when it is supported by both ends 1. Also note that there is no real call setup, so in some cases there is no call setup delay. The initial Vo users have the advantage of lower communication costs, especially for long distance calls. Vo makes it possible to talk to the other side of the world at the cost of a local call (under the assumption that the other party is also connected to the internet. The cost benefits for the user are becoming less important, since also the classical telecom operators are dropping their prices. Some operators start using Vo internally in the network with gateways to the PSTN network for global connectivity and share the cost savings with their subscribers. Another benefit for the users is the flexible introduction of new services. The network provides the transport of data but is a dumb network, the services are in the end-terminals. You only have to upgrade the end-terminals to install a new service like video conferencing. Note that flexible mechanisms (like java) have to be installed in end-user devices for automatically downloading or upgrading of terminal software to make this true for the common user. Introduction of Multimedia applications will be possible with the introduction of higher bandwidths and better QoS in UMTS. Some of the benefits of using Vo are summarised in the following list: For Operators: - Efficient use of network resources - Packet switched circuit switched; no performance cost for circuit (call) setup - Reduce number of networks; only one network to manage and maintain - Multiple providers few operators; resell transport capabilities to service providers - Lower cost of IT equipment (IT equipment vs. telecom equipment) 1 This can result in a global bandwidth gain of approximately 4, (using additional techniques like silence suppression).

3 For Users - Multimedia support - International calls at local tariff - Flexible introduction of new services (only upgrades at the end-users); users are in control of the services they want; no need to wait until the telecom operator of today introduces it. - Allows end-users to program the behaviour of telephone-like services. - Possibility of surfing on the Internet and receiving a call at the same time. - Adjustable quality by selecting the appropriate coding in function of the available bandwidth This comes of course at a certain price: - lack of QoS in packet based networks - interoperability with between the different implementations and protocols - scalability (e.g. H.323) - how can an operator bill this; no billing in. - security There are several scenarios for introducing Voice over in the mobile network. We investigate 2 possibilities: using Vo as end-to-end service and using Vo purely internally in the mobile network. The following calculations give an idea of the necessary bandwidth for Voice over applications. The calculations are made for both the full rate GSM codec and the half rate GSM codec, other codecs can of course also be used. Full rate GSM codec: A packet is generated every 20ms resulting in a bandwidth of. This is the bandwidth necessary for Vo with a 13 kbps GSM codec. Half rate GSM codec: Resulting in a total bandwidth of for a half-rate GSM codec. This big overhead is caused by the size of the headers (> 50% of total packet size). Sending empty packets (only // headers) every 20 ms already consumes. These figures indicate that from the bandwidth usage perspective, standard Vo is not feasible. To reduce the overhead of the headers we can use header compression on link-by-link connections. The combined // header compression reduces the size of the headers to 4 bytes with checksum or 2 bytes without checksum. This technique can be used in combination with PPP tunnels or implemented in the SNDCP protocol (GPRS), thereby reducing the header overhead from 16 kbps to 0.8 kbps. Using this technique we arrive at for a full rate codec. When we use Vo for an end-to-end telephony call in a mobile network, we have to use a data connection. The data transported over this connection is not checked by the network; it is transported transparently. The mobile operator is not aware of the type of data transported. All Vo devices (gateways, gatekeepers, )

4 are also located outside the mobile network, not under the control of the mobile operator. The user who wants to use Vo has to setup a GSM data call to his ISP, this implies transporting packets over the radio interface. An intelligent end-user might think of using this solution for the same reason Vo was initially invented: making international calls at the cost of a local data call. This is not realistic not only due to the maximum bandwidth of 9.6 kbps (14.4 kbps) for a GSM data call. But also the big delays for GSM data calls (typical values around 200 ms) make them unusable for real-time services. Note also that implementing an // stack plus codec in the terminal demands a lot processing power from the mobile station, making the mobile phone too expensive for common users (who would benefit the most). GPRS allows higher bandwidths than GSM but the bandwidth is shared between different users. A possible solution for the QoS problem is to give the Vo (real-time) packets a higher priority. Using Vo over the GPRS radio interface is (still) less efficient than GSM. One reason to use Vo on the radio link is to avoid gateways in the path (complexity, quality loss, extra delay), end-to-end also make it possible to introduce multimedia applications. A special case of Vo in GPRS is for class C busy terminals (class C = only GPRS). Users currently attached to the GPRS network can then still receive voice calls. The user gets a notification on his terminal and can choose to accept or reject the call. If the incoming call comes from the PSTN we need a Vo gateway at the edge of the GPRS network [Figure 1]. BSS SG SN GGSN GW CS network Protocol stack in the MS Public In UMTS the GPRS network will be interconnected to the UTRAN providing higher bandwidths. However, QoS and especially delay still stays an important issue. In this scenario we put a Vo gateway at the edge of the GSM network, under control of the mobile operator. In the mobile network a normal GSM voice call is setup, conversion to packets is done at the edge of the network. The mobile phone user does not see the use of Vo at all, implying that no changes to the mobile phone itself are needed. We name these scenario s non-transparent since the use and implementation is completely under control of the mobile network. The benefit here is mainly for the

5 operator, using the cheap internet as an alternative for an expensive (both in terms of equipment and management) backbone for PSTN. G W Public BSS MSC MSC CS network This is of course again at reduced quality. Note that this quality reduction can only be investigated in practice; quality of service is generally provided by over-dimensioning the network. A mobile operator could offer this to his subscribers as a special service (cost <-> quality) for international calls or do this completely transparently. The decision to transcode GSM to Vo could be made on a number of criteria: knowledge of the quality of transmission to a certain country (end destination) type of the called party (a Vo end-user) cost of the link to a particular country loading of the normal core network subscription to the service The last scenario we discuss in this paper, is the use of Vo between two MSCs, for a voice call between two mobile terminals [Figure 3]. Both anchor MSCs are interconnected through an network. This could be especially interesting for long distance calls. Operators could use this architecture to provide some load balancing between the PSTN network and the network. In order to be able to use the Vo backbone we need to place Vo gateways at the MSCs. Handovers that would involve a change of the gateway (at either end) cause specific problem out of the scope of this paper. Switching gateways would cause synchronisation problems since it is at the gateway that the timing information is added to the packets.

6 G W G W BSS T T MSC MSC C C BSS CS network Note that ideas from the work done on (TFO) could be used in this architecture. In transcoder free operation the GSM speech is not transcoded to PCM between the BSC and the MSC, transcoding causes additional quality loss and extra delays. Before we can provide end-to-end Vo solutions - packets over the radio interface- in a mobile environment there will have to be a substantial increase of bandwidth on the radio link and better QoS guarantees. Also implementing an // stack plus codec in the terminal demands too much processing power from the mobile station, making the device expensive. GSM is specially designed to make optimal use of the radio interface in term of bandwidth. So, it will be hard to invent new schemes that make better use of the available radio resources and if they were found they would probably introduced in the GSM standard as a new coded. The current QoS support for GSM data connections is insufficient to support real-time data traffic. In GPRS there is some more support for QoS, but QoS profiles are associated to a complete PDP context with no possibility to differentiate between the different flows within one PDP context. This will be improved for UMTS. If Vo will be introduced in the future, it is most likely that it will be first introduced in the network. This might offer the operators some savings on the cost of long distance connections. A few possible scenario s and network configurations were presented.