SIP Signalling Resources in 3GPP IP Multimedia Subsystems Alexander A. Kist and Richard J. Harris RMIT University Melbourne BOX 2476V Victoria 3001, Australia Telephone: (+) 61 (3) 9925-5218 Fax: (+) 61 (3) 9925-3748 Email: fkist,
[email protected]
Abstract— The Session Initiation Protocol (SIP) will be used as the signalling protocol in the IP Multimedia (IM) Subsystem of the 3rd Generation Partnership Project (3GPP) UMTS networks. Using the protocol on a large scale to provide equivalent telephony service requires Quality of Service (QoS) observations. This paper discusses SIP signalling resources and their impact on the QoS. The first part summaries observations in conjunction with signalling resources. The second part introduces methodologies that enable the dynamic resource allocation for SIP overlay networks. The third part shows areas where further study is required.
I. I NTRODUCTION Signalling traffic is considered to be an important type of network traffic and lost signalling messages or congestion of the signalling network can have a devastating impact on all services that rely on signalling sessions. Providing QoS for signalling is therefore one of the critical tasks in implementing session related services. The Session Initiation Protocol (SIP) is used as a signalling protocol in an IP environment. It performs user location, session establishment, session management and participant invocation. The SIP protocol is defined in RFC 2543 [1] and a new version of the specification is discussed as an Internet Draft [2] (work in progress). There are several publications available that provide an introduction to the SIP protocol (e.g. Schulzrinne/Rosenberg [3]). In 3GPP, the SIP functions are located in the IP Multimedia Subsystem (IMS). The currently applicable 3GPP specification is [4] (work in progress). Mobility aspects and configuration issues like the assignment of the IP address to User Equipment (UE) are not the focus of this paper. The 3GPP specification allows two possible IP context implementations: Signalling can use a separate IP signalling context or it can use the general purpose IP context. QoS methodologies are required to enable service provision for the overlay signalling network especially in the latter case. Signalling information has to be transported most efficiently in the network since all delays result directly in session handling and network management issues like bearer reservation etc. Performance modelling and evaluation can be considered as one of the main steps towards an integrated QoS scheme for signalling. X. Xiao and L.M. Ni discuss in [5] the possibility
of providing QoS for the Internet. Some of their general observations can be applied to the 3GPP network. A successful QoS scheme for SIP in 3GPP networks has to span over all involved layers. These are: the application layer with the SIP protocol, the transport layer, using by default the UDP protocol, and the network layer using the IP protocol. For the SIP layer, this involves the direction of traffic to areas of the network that are not congested, as well as load balancing schemes and the development of backup methodologies. The transport layer has to be either dedicated to the signalling or a QoS scheme like Diffserv [6] has to be implemented. The network layer has to consider traffic engineering methodologies to avoid congestion, for example MPLS [7] and local repair methodologies. This paper is mainly concerned with a QoS scheme for the SIP overlay network. Section II discusses signalling resources and their implications for Quality of Service. The situation where a general-purpose network is used for the signalling is outlined in detail. Possible planning methodologies to enable QoS signalling are introduced for general-purpose networks. Section IV shows areas where further study is required. II. S IGNALLING R ESOURCES The signalling transport network will most likely be IP based. Two general network configurations are possible: The signalling can use a dedicated or a general-purpose network. This paper focuses on the general-purpose network situation. This section discusses different aspects: the signalling network traffic, SIP overlay network connections and the general-purpose network. Network Traffic: There are considerable differences between conventional Internet traffic and signalling traffic. The Internet accommodates a traffic mix of different applications and services. The TCP and UDP protocols are used as transport protocols. The dominant connection type in today’s networks is TCP (e.g. [8]). Connections extend from short-time connections of a few milliseconds to long-term connections of many hours. The signalling network will be different. UDP traffic will be dominant. Beside the general network structure of IP signalling networks, the traffic encountered in such networks will be closely related to existing packet switched signalling
networks like Signalling System No.7 (SS7) [9]. General considerations (e.g. [10], [11]) and methodologies (e.g. [12]) originally introduced for SS7 can serve as a basis for the modelling of session signalling traffic in an IP environment. The history of SS7 operation shows for example that low link utilization is necessary to cope with the burstiness of signalling traffic. SIP overlay network links are the application layer connections that link the SIP nodes. The QoS for SIP messages can be classified by two parameters: the delay and the message loss probability. The delay is a sum of fixed delays which include the transmission delay at the network nodes, the propagation delay at the network links and variable delays including queueing and processing delays in the network nodes. On highspeed links, the propagation delay is the dominant fixed delay parameter and it can be assumed to be independent of other factors. The variable delays are dominated by the queueing delay. Queueing delays are influenced by utilisation on the transport network, the average flow and higher order statistical properties of the packet arrival process. Link layer retransmissions add to the queueing delay. The SIP link delay is, therefore, mainly the sum of propagation delays on the bearer links and queueing delays that occur on the transport network. Message loss is caused by bit errors on the transport network links and dropped messages in overflowing queues. Where the first depends on the message size and the second on the utilization and statistical properties of the message arrival process. If the end-to-end delay exceeds a timer value the message is also considered to be lost. General-Purpose Network: Different network services will use general-purpose networks at the same time. These services have different requirements for quality of service. In such a case, network traffic has to be prioritised. To implement quality of service guarantees, networks will use quality of service schemes. Two general approaches are under consideration. The integrated service scheme (IntServ) [13] is a reservation-based mechanism. A dynamic signalling protocol reserves resources for individual flows. For large networks this approach is not scalable (e.g. [5]). The second approach is the use of Differentiated Services (DiffServ) [6]. It uses well-defined classes which are provided with differentiated services. At the edge of the network, routers classify packets on the basis of traffic classes and service level agreements. Beside the traffic class, the packets are marked in or out of profile based on predefined subscriptions of users. The routers within the network process the packets according to the traffic class. In the case of congestion, the out of profile packets are dropped first. In a well-engineered network, the performance within one high priority traffic class should be close to the performance expected from a dedicated network even if DiffServ provides no absolute guarantees. This assumes that the coupling between the different service classes is minimal. It is likely that network routers will use advanced queue management schemes like adaptive Random Early Detection (RED) [14]. If these systems are implemented correctly and the network is sufficiently engineered the queueing delay
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will be only a fraction of the propagation delay. This can simplify the delay forecast considerably. To avoid disruptions in the high priority signalling traffic a careful dimensioning effort for the overall transport network is required. Where Internet services using TCP can reduce the traffic sent into the network, in the case of congestion, this situation is not acceptable for signalling. For optimal performance of signalling it is required that no packets are lost or, more realistically, a minimum number of packets are lost due to queue overflows. Where, for many multimedia applications, the loss of packets is acceptable, the loss of signalling messages is more problematic. A signalling packet is more precious since the information per packet on a larger scale and the impact of the loss of this packet is considerably higher than the loss of information media packets carry in a voice session for. Traditional signalling networks show that the signalling traffic is bursty. To over-dimension or oversubscribe the network resources in general purpose QoS networks is one way of taking the burstiness into account and avoid the loss of signalling messages due to buffer overflows. The over-subscription has to be of an order that practically all signalling packets are marked as in-profile. In Differentiated Services schemes, the unused signalling resources due to the low link utilization can be used by low priority services. If the use of these resources is limited to out-ofprofile packets by the other services, this causes no disruption of the signalling messages even in the case of moderate congestion. In case of congestion or when signalling requires resources, the out-of-profile packets will be dropped first. The usability of oversubscribed resources by low priority traffic is one of the advantages gained through the use of general-purpose networks. In designated signalling networks these resources are lost. Another main advantage of a general-purpose network is the possibility of dynamic resource subscription. Traditional signalling networks require a high effort in capital for the building up of reliable signalling networks. Resources have to be available for the maximum number of users including reliability and back up considerations during the installation of the network. These resources are only assigned to signalling. Generalpurpose networks, on the other hand, require similar attention on a larger scale for all services. The assignment of resources to single services in such networks is not bounded by physical means. The resources can be assigned on a short timescale. This can be part of the network design or it can be part of a dynamic resource reservation process. If it is possible to estimate the required signalling resources, for example on the basis of “number of users registered in network nodes”, the resource assignment can be adapted dynamically. This process will not be session based, but depend on a larger timescale like the duration of user registration or general time of day estimates. It is obvious that the resources have to be available to apply such schemes. Taking into consideration that the signalling traffic will be a fraction of the traffic caused by other services, the signalling will place no critical burden on the network resources.
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7 Protocol Parameters
Topology
Delay Loss Boundaries
3 Max number of Users, Functional Routing Requirements, etc.
Node Parameters 2 e.g.. Number of Current Users, Load, etc.
5 Optimisation Methodology
Overlay Network Parameters
8 Flow Model
Mean Flows
Resource Subscription
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9 TSpec
Overlay SIP Links
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Fig. 1. Modelling Concept
III. DYNAMIC R ESOURCE S UBSCRIPTION This section discusses a methodology to estimate signalling resources required by the SIP overlay network. The overlay network consists of two items: SIP nodes which are specified by the number of users, the processing delay etc. and the SIP overlay network links which are defined by traffic specification parameters (T-SPEC). SIP overlay links are virtual connections between two SIP nodes defined by the resource subscription. The underlying IP network provides in general full connectivity. The here discussed method uses available information about the SIP nodes to calculate the required SIP overlay network links resources. Subsequently the SIP overlay network subscribes these resources form the transport network. Figure 1 depicts the dynamic resource subscription scheme. Depending on the dynamic input data this scheme can also be used as a planning and optimisation methodology. The Network Topology (1) function provides the basic static data of the overlay network. This includes the functional specification of SIP nodes, required connectivity between certain SIP nodes and other static information like the maximum number of users served per node etc. The Node Parameters (2) provide dynamic information about the current state of the SIP network nodes like the number of currently registered users etc. The Link Parameters (3) provide similar dynamic information about the current state of the virtual connections. This includes the currently available resources and loss probability. The Optimisation Methodology (4) is the main function in this scheme. It uses the network topology information, the current node and link state, the overlay link information and delay bounds to make qualified routing and subscription decisions. Messages on the SIP layer are routed to pass certain functional nodes specified by the technical 3GPP specifications. Certain nodes hold registration or session state information. It is therefore essential that these nodes are part of the route. The Flow Model (5) function uses the overlay network parameters and the SIP Protocol Parameters (6) as well as the Delay and Loss Boundaries (7) to calculate the message flows between SIP nodes. Signalling messages sent using the SIP protocol pass through intermediate SIP nodes in such a way that the message size grows as a result of additional data being added to the message. A flow model is required to get an appreciation of the size of these expected sig-
nalling flows. A model is introduced in [15]. Once the required flows between the SIP nodes are known the traffic specification parameters have to be estimated. The Resource Subscription (8) function utilises a model to calculate the peek rate and the bust tolerance using the mean rate, the maximum message size and the Delay and Loss Boundaries. The size of the actual subscribed amount depends on the Service Level Agreement (SLA) and the subscription ratio (subscribed resources/calculated resources). The actual traffic subscription defines the Overlay Network SIP Links (7). These parameters are also feed back to the Optimisation Methodology. This methodology compromises a feedback system. It starts off with the topology and predefined usage values. Using this values the expected flows on the SIP overlay network links are calculated. In the next step the required reservation is considered and fed back into the optimisation and routing methodology. If the scheme is used for dynamic resource subscription the time scale is important. Since many message routing decisions are registration based rather than session based the resource subscription time should be of a similar scale. The subscription ratio has to be high enough that possible fluctuations during the subscription time are well within the available resources. If the scheme is used for dimensioning purposes the dynamic input values are set to maximum levels e.g. the maximum amount of users per note and the traffic specifications are calculated. IV. F URTHER S TUDY There are three topics that require specific attention in further research efforts. These are discussed below: SIP Link Model : SIP nodes are connected with SIP links on the application layer. These connections will use Differentiated Services networks. Many studies have discussed the effects of TCP traffic in Differentiated Services networks. A current example is C.Barakat and E.Altman [16]. At the edge of the network, flows are monitored and classified on the basis of rate and burst. The SIP link model has the model UDP traffic in Differentiated Services networks on the basis of inputs from the SIP layer flow model. Optimal Routing on the SIP Layer: Routing decisions on the SIP application layer are done during the registration of users or during session initiation. After the process of registration or after the session initiation is completed, certain nodes hold state information. Subsequent messages are required to pass these nodes. Changes of the message flow routing for subsequent requests are only possible when different nodes of the same type hold the state information e.g. backup notes. Issues related to routing and therefore to resource assignment are load balancing, avoidance of congestion and failure scenarios as well as optimisation considerations like delay and message loss. Factors with possible impact on routing decisions are: local policy, security issues and load of connection and/or servers. Details of this process are for further study.
Transport Protocol and SIP: The SIP protocol is transport protocol independent. Therefore, it needs to incorporate its own end-to-end reliability mechanism. In particular the SIP extension known as “Reliability of Provisional Responses” introduces an additional reliability mechanism for the provisional response in a SIP call flow. The TCP protocol is supported, but one of the major disadvantages of TCP for signalling is the protocols’ long connection set up time. UDP the default transport protocol has the disadvantage of its unreliability. In particular, this can cause problems in 3GPP with the high number of intermediate links and the fact that the reliability is end-to-end. The new Stream Control Transmission Protocol (SCTP) [17] could be an alternative with considerable advantages for the session setup times and throughput. Internet Draft “draft-ietf-sip-sctp03.txt” [18] (work in progress) discusses the SCTP protocol in conjunction with its use in SIP. It suggests that most of the benefits of SCTP occur under loss conditions. Under zero loss conditions SCTP and TCP should perform similarly. The situation in 3GPP domains is different from the situation in general Internet environments. As mentioned above, the TCP protocol is not the preferred choice but the SCTP protocol could allow a similar performance for SID than UDP but provides network layer reliability. The following items can extend the catalogue of possible advantages in 3GPP domains using SCTP: Fast Retransmit: SCTP incorporates a reliability mechanism that quickly detects the loss of packets. This is faster then the in SIP incorporated reliability mechanism. Beside the general benefit for normal SIP connections this offers especially an improvement in 3GPP since the standard connection incorporates up to eight intermediate proxies and SIP’s reliability is for most connections end-to-end. Streams: SCTP supports multiple streams. Between CSCF nodes within the 3GPP SIP core network many transactions occur simultaneously. If these nodes establish permanent connections, SCTP can provide similar performance than SIP over UDP. Streams within one SCTP connection realize the single SIP transactions. The reliability of a TCP connection is also provided. This is probably one of the main benefits of SCTP in a 3GPP environment. Multihoming: Multiple IP addresses for one host adapter can be associated with a single SCTP connection. If one destination gets unreachable the connection can use the second interface. Between CSCF nodes with many SIP connections this can provide considerable performance improvements over the possibility of routing SIP messages on the basis of the SRV procedure defined in [1]. Several CSCFs maintain state information for existing SIP connections. In the case of network failure, the application layer has to reroute the SIP messages. The failure procedure might include data synchronization and network activity to request SRV records. Using the SCTP multihoming feature the network failure will be unnoticed on the application layer.
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V. S UMMARY
This paper provided an overview of Quality of Service issues arising through the use of the Session Initiation Protocol on a large scale in 3rd Generation Partnership Project IM Subsystem, were the IMS is used to provide equivalent telephony services. The situation for the general-purpose network was outlined in detail. A dynamic resource subscription scheme was introduced. This scheme also enables the development and implementation of planning tools for performance modelling and optimisation of the QoS network used for the signalling in 3GPP IP multimedia subsystems. VI. ACKNOWLEDGEMENTS The authors would like to thank Ericsson AsiaPacificLab Australia for their financial assistance for this work. R EFERENCES [1] M. Handley, Henning Schulzrinne, E. Schooler, and Jonathan D. Rosenberg, SIP: Session Initiation Protocol, March 1999, RFC 2543. [2] Rosenberg, Schulzrinne, Camarillo, Johnston, Peterson, Sparks, Handley, and Schooler, SIP: Session Initiation Protocol, IETF, February 2002, Internet Draft draft-ietf-sip-rfc2543bis-07.ps (work in progress). [3] Henning Schulzrinne and Jonathan D. Rosenberg, “The Session Initiation Protocol: Internet-Centric Signaling,” IEEE Communications Magazine, pp. 134–141, October 2000. [4] 3rd Generation Partnership Project, IP Multimedia (IM) Subsystem Stage 2 (Release 5), July 2001, 3GPP TS 23.228 V5.1.0. [5] X. Xiao and L. Ni, “Internet QoS: A Big Picture,” IEEE Network, March/April 1999. [6] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, and W. Weiss, An Architecture for Differentiated Services, IETF, December 1998, RFC 2475. [7] E. Rosen, A. Viswanathan, and R. Callon, Multiprotocol Label Switching Architecture, IETF, January 2001, RFC 3031. [8] H.Sawashima, Y.Hori, H.Sunahara, and Y.Oie, Characteristics of UDP Packet Loss: Effect of TCP Traffic, INET, 1997, http://www.isoc.org/isoc/whatis/conferences/inet/97/proceedings/F3/F3 1.HTM. [9] Ronald A. Skoog Adi R. Modarressi, “Signaling System No. 7: A Tutorial,” IEEE Communications Magazine, pp. 19 – 35, July 1990. [10] ITU, ITU-T Recommendation E.713, October 1992, Control Plane Traffic Modelling. [11] ITU, ITU-T Recommendation E.733, February 1996, Methods for Dimensioning Resources in Signalling System No. 7 Networks. [12] R. Girao et. al., “Dimensioning Signalling Links (SS7) in INs,” April 2001, http://www.co.it.pt/conftele2001/proc/proc.html. [13] C. Bormann, Providing Integrated Services over Low-bitrate Links, IETF, September 1999, RFC 2689. [14] Sally Floyd, Ramakrishna Gummadi, and Scott Shenker, “Adaptive RED: An Algorithm for Increasing the Robustness of REDs,” August 2001, Under Submission, http://www.aciri.org/floyd/red.html. [15] Alexander A. Kist and R.J. Harris, “A Simple Model for Calculating SIP Signalling Flows in 3GPP Networks,” In Proceedings of the Second IFIP-TC6 Networking Conference 2002, Pisa, May 19-24, May 2002. [16] Chadi Barakat and Eitan Altman, “A Markovian Model for TCP Analysis in a Differentiated Services Network,” In Proceedings of First COST 263 International Workshop,QofIS 2000, pp. 55–67, September 2000. [17] R.Stewart et.al., Stream Control Transmission Protocol, IETF, October 2000, RFC 2960. [18] Jonathan D. Rosenberg, Henning Schulzrinne, and Camarillo, SCTP as a Transport for SIP, IETF, June 2002, Internet Draft draft-ietf-sip-sctp03.txt (work in progress).
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