Towards a User-Centric and Quality-Aware Multimedia Service Delivery Implementation on IP Multimedia Subsystem Chitra Balakrishna, Khalid Al-Begain
[email protected],
[email protected]
Abstract IP Multimedia Subsystem (IMS) originating from the 3GPP is pivotal on the road to next generation all-IP networks. Multimedia applications are potential drivers for the adoption of next generation IP multimedia subsystem (IMS) and could be a decisive factor in its deployment. Innovative multimedia services are expected to generate revenue to the operators and justify their investment in IMS. And, for multimedia services to be widely accepted by the end users, guaranteed and adaptable quality of service provision across heterogeneous access networks and heterogeneous device types is mandatory. In this paper, a generic, adaptive, controlled service delivery framework implementation for multimedia applications on IMS architecture is proposed. The framework is designed to be user-centric, adaptable to terminal mobility and session mobility with minimum session disruption enabling a better end-to-end QoS. It is at the session-level, completely user-driven and does not account for the network-level QoS. It employs the Session initiation protocol (SIP), which is the signaling protocol for the 3GPP specified IMS. The purpose behind the framework is to evaluate the provisioning of quality-adaptive multimedia services on IMS network and analyze the performance with respect to delay induced by quality adaptation procedures during session and terminal mobility which might contribute towards the end-to-end quality of service delivered to the users. Keywords: IMS, B2BUA, Session Mobility, Adaptive Quality
1. Introduction Internet is the largest and the most popular public data network based on packet-switched technology. With the worldwide deployment of circuit-switched cellular network, services offered are not limited to voice alone. Demand for multimedia services and internet-like applications by the mobile users are increasing. It is only natural to expect future network architecture to bridge this divide between circuit-
switched cellular network and the packet switched Internet and consolidates them into a single network providing a common set of services to the users anywhere, anytime. The 3GPP IMS is expected to serve this very purpose by creating an accessindependent environment and making the mobile Internet paradigm a reality. IMS is only the core network, which is limited to the control layer and relies on an adequate service delivery platform in the application plane to cater to the user’s demands of innovative multimedia services and to provision appropriate session-level quality of service [1]. We propose a SIP [2] based implementation of a multimedia service delivery framework on IMS core network, to ensure service quality that is adaptable to terminal mobility and session mobility and minimizes delay induced by mobility and by quality adaptation procedures. The multimedia service considered is of Server-to-User type, where one or many users access a set of multimedia streaming service from a 3rd party server or internet. Rest of this paper is organized as follows, IMS architecture with research scenario is briefly discussed in section 2, next generation mobile user demands and quality requirements and related work in section 3, the proposed SIP-based multimedia service delivery implementation is detailed with an overview of the operation in section 5, and section 6 concludes the paper with a brief discussion on the contributions of the framework and future work.
2. IP Multimedia Subsystem Architecture IP Multimedia Subsystem is an architecture standardized by 3GPP to offer a common core for next generation networking technologies [3]. In order to achieve access-independence and smooth interoperation between the circuit-switched and packet switched networks, IMS conforms to the IETF standards. IMS marks a transition from vertical service architecture to a horizontal one. It uses layered architectural design, where the transport and bearer services are separated from the signaling and session management services. Layered approach aims at minimum dependency between layers. It forms the core
network constituting the control plane and relies on an adequate application plane for multimedia service provisioning. Figure 1 shows the global view of the IMS architecture for which the multimedia service delivery framework implementation is proposed. It shows the research scenarios the framework considers, such as terminal mobility and session mobility. The architecture consists of various components that constitute the connectivity plane, control plane and the application plane. The framework for the multimedia service delivery is proposed to be implemented in the application plane using a SIP Application server and Media server from the control plane . Streamin g Server
3rd Party Providers
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P-CSCF PDF
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• Proxy-CSCF (P-CSCF) P-CSCF acts as the first point of contact for call signalling coming from the User Equipment (UE) to the IMS network. From SIP[3](rfc) point of view, the PCSCF is another outbound/inbound SIP proxy server. All the requests and responses initiated by or destined to the IMS terminal will go trough the P-CSCF. PCSCF does not change once the session is established. • Interrogating-CSCF (I-CSCF) I-CSCF is a SIP proxy located at the boundary of the IMS network and acts as an entry point for SIP signalling coming from another network. I-CSCF is usually located in the home network1 and is responsible for assigning an S-CSCF to the subscriber. • Serving-CSCF (S-CSCF) S-CSCF is the central node of the signalling plane. The S-CSCF is essentially a SIP server acting as a SIP registrar. It also carries out the call/session and accounting control for a given subscriber. • SIP Application Server Application plane contains application servers, which provide the end-user service logic. IMS provides a variety of telephony and non-telephony application servers to cater broad range of voice and multimedia services. • Media Server Media Server performs a specialized and a specific role in the IP network, by providing real-time media packet processing and transcoding for rich-multimedia next generation applications. SIP server sends SIP signals to the media server requesting a service from the media server and sets up the media stream between the media server and SIP end terminals.
GSM/GPRS/3G
3. Next-Generation Requirements Session Mobility Terminal Mobility
Figure 1: Global View of IMS Architecture The components that are relevant to the framework are discussed in the following paragraph. Three types of CSCF are defined, each type providing different functionality.
Mobile
User
Given that today’s network architecture is evolving towards a converged, centralized and managed IMS platform, user’s service and quality requirements are also changing accordingly. From [4], it can be assumed that the next generation networks and services would be defined by mobility and seamless access to rich multimedia services across varying device types and varying access technologies.
The above user needs bring with them a complex QoS demand, such as adapting QoS to user’s context and preference. Therefore the proposed service delivery framework is designed to be adaptive, quality-aware and user-driven. Seamless service across varying access technology is achieved by supporting vertical handoff at the control plane. Mobility management protocols in general are responsible for supporting services seamlessly across heterogeneous access networks. A lot of research work is directed towards solving the vertical handoff problem in IP-based networks with optimum QoS performance. Based on the layer in which they operate, these mobility management protocols are classified as those operating on networklayer [5], transport-layer [7][8] or application layer [10][11]. Table 1, compares the mobility management protocols at different layers. Operating Layer Network Layer
Protocol
Advantages/ Limitations
Mobile IP and its derivative
* Triangular routing problem. * Changes needed in IP stack [6]. * Transport protocol cannot handle the change in IP. * Timeouts suffered by TCP affects performance. * Depends on IPSec, adding to overheads and delay. * Not suitable for fast handoffs. * Maintains end-to-end semantics, suitable for heterogeneous access networks.
Transport Layer
TCPextensions [7], SCTP[9]
Application Layer
SIP
Table 1: Comparison of Mobility Management Protocols The dependence on the access network decreases as we move up the protocol stack. Application layer protocols are transparent to network layer and lower layer characteristics. They maintain true end-to-end connection semantics and are hence most suitable for heterogeneous access network environments [12][13]. Amongst the application layer protocols, SIP is standardized by the IETF and also supports terminal mobility and personal mobility along with network
mobility and is most appropriate protocol for mobility management. Seamless service across multiple access devices is termed as session mobility [10][14], where a user maintains an on-going media session while changing terminals. Although SIP is universally accepted as the most suitable protocol to address mobility issues related to QoS and adaptation for multimedia services to be resolved.
3.1 Quality of Service requirements for Multimedia Applications The framework is proposed for Server-User Multimedia service, where traffic is non-real-time and delivery could be unicast. Fundamental parameters affecting QoS for non real-time multimedia services are: end-to-end delay, delay variation or jitter and packet loss. The above parameters depend on the network condition on the path carrying the data traffic. But, terminal mobility handoff delay and session mobility handoff delay and delay incurred during the quality adaptation process to adapt to changing device type and network types, does contribute to the end-to-end delay. A large handoff delay could cause considerable packet loss affecting the overall user perceived QoS for multimedia services. Analysis of handoff delays [15] show that delay as low as one second could have considerable effect on the multimedia service continuity and therefore quality when devices are connected to wireless access networks with low bandwidth [16]. Implementation proposed in the framework aims to minimize the application-level handoff delay due to mobility and quality adaptation thus contributing towards the end-to-end QoS.
4. Problem description Primarily, terminal mobility across different access network technologies is supported by mobile terminals with dual/multiple access network interfaces. Upon acquiring the new IP address during mobility and completion of network-level handoff, SIP takes over to perform the application-level hand off to complete the terminal mobility management procedure. Session mobility allows a user to maintain a media session while changing terminals. It is mostly initiated by the terminal and executed at the SIP controller (CSCF) in the operator’s network. SIP as per the IETF specifications handles terminal mobility at the application plane using Re-Invite
method.[17]and session mobility using the REFER method[18]. Potential issues that need to be resolved to handle session and terminal mobility are 1. Transfer Delay – time for application setup, time for media buffering. 2. Media Disruption – frame loss / packet loss 3. Different Device Capabilities – media codec, display size, buffer capacity. 4. Different Access network capabilities – Bandwitdh, Max Bit rate etc. Our framework implementation aims to resolve few of the above mentioned issues. The design requirements for the framework, drawn from the discussion above are listed: 1) Quality of the multimedia session adapted to the access network bandwidth/bit-rate. 2) Quality of the multimedia session adapted to device type in terms of codec supported by the device. 3) Minimize the delay due to mobility. 4) Minimize the delay due to quality adaptation.
5. Framework Implementation Operational Overview
and
Figure 2, shows the components of the IMS architecture, which are part of the quality-aware multimedia service delivery framework. SIP application server in the application plane together with the media server in the control plane provide the necessary adaptation, to the user accessing any streaming application from the third party providers or the internet, as the user goes mobile across multiple access technology or the streaming session moves across multiple devices. SIP is the protocol used for the initiating, modifying, termination and adapting the multimedia streaming session between the server and the end user, while a markup language such as MSCXML [19] is used by SIP to control the media server.
5.1. SIP Headers and SDP parameters used for Adaptation 1. P-Access Network Info Method [20]: This header is generated by User agents on the mobile device as part of SIP request/response to convey to the serving SIP application server about the access network the device is currently connected to. The SIP A/S uses this information to provide optimized and adapted service quality for the end terminal based on the access
technology bandwidth capabilities. This header is crucial for providing the terminal mobility support and adaptation. SIP Application Server
SIP (B2BUA B2BUA) SIP Messag e
Media Server
MRFC MRFP
Figure 2: Framework Components 2. Session Description Protocol (SDP) [21]: SDP parameters describe the multimedia sessions and are sent as part of the SIP messages to the SIP A/S. Some session level parameters that are of relevance are: The type of media (video, audio, etc.) - The transport protocol (RTP/UDP/IP, H.320, etc.) - The format of the media (H.261 video, MPEG video, etc.) - The frame rate in frames per second SIP A/S uses the above mentioned parameters to provide session adaptation, primarily during session mobility.
5.2. Framework Components Back-to-Back User Agent (B2BUA) [2]: SIP A/S by default acts as a SIP proxy capable of only routing SIP messages. To be able to control the SIP sessions, alter the SDP parameters for adaptation and to be able to terminate calls like user agents, SIP A/S is implemented as. B2BUA in our implementation acts as the 3rd party call (TPCC) controller, enabling creation and control of the multimedia session between the enduser and the streaming server via the media server, which acts as the RTP replicator and transcoder. SIP message parser: It parses the relevant SIP headers and SDP parameters from the SIP messages and generates new SIP message with the above parameters to be sent upstream to the streaming server, enabling media and session negotiation between the Streaming server and the SIP A/S. Media server [22]: It has two functional units controller (MRFC) and processor (MRFP), thus separating the functions of media control from media
processing. SIP A/S controls the media content by interacting with the media controller and the media processor, processes the media on behalf of the SIP A/S, provides transcoding to the end-user and in our implementation, it also provides media mixing features as in a conference. The B2BUA in combination with the media server acts as the focal point of the multimedia service delivery and adaptation. The B2BUA acts as the conference focus by connecting the end-user and the streaming server to the media server treating them as independent conference legs, the media server receives media content from the streaming server, mixes and transcodes the content as per the end-user condition, taking into account the access network and the device type the user is currently using. This information is fed to the media server by the SIP A/S-SIP Parser combination. Media server adapts the media content to terminal mobility by encoding the media at a rate which matches the bandwidth capabilities of the access technology. Media content is adapted to the session mobility across a different device-type, by encoding in a format supported by the end-device, also processing the media to suit the display capabilities of the end-device in terms of frame rate and resolution. SIP application Server is the brain of this architecture which contains the application logic, although the actual content resides elsewhere, while the media server provides the muscle by enabling playing and recording of media streams, transcoding between various formats, audio-video synchronization. The media server is controlled by the app server using sip signalling and a markup language such as VoiceXML (VoXML).
streams the content it receives from the streaming server to the end-terminal. Stage 2: When the terminal goes mobile and connects to a different access network and acquires a new IP address, it issues a re-invite message to the SIP A/S with its new IP address, new access network type via the p-access network info header and new session requirements [Figure 4]. SIP A/S forwards the re-invite to the media server. The media server encodes the media at a rate that is suitable to the new access network bandwidth capabilities, and starts streaming media to the new IP. It disconnects the media to the old IP address. Note that the session renegotiation happens between the SIP A/S and the end terminal and the quality adaptation is performed at the media server, and not between the sender and the receiver. Note that this renegotiation happens between the UE and the SIP A/S in the same provider network. This minimizes the delay incurred due to negotiation and adaptation. Also the third party streaming server may not have transcoding capabilities to perform adaptation. Both SIP A/S and the media server are part of the operator’s core network and hence would have a better control over the media content. UE
S-CSCF INVITE
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INVITE 200 OK
200 OK ACK
ACK One-way INVITE 200 OK SDP2 INVITE 200 OK SDP3 ACK SDP3 ACK ACK One-way RTP One-way
The operation of the framework is split into three stages; session establishment, Session adaptation due to terminal mobility and session adaptation due to session mobility. Stage 1: For purpose of simplicity, IMS registration and the call flow through P-CSCF and I-CSCF is not shown. S-CSCF is what connects the multimedia session initiated by the end-user to the SIP A/S, where the application logic is present. Figure 3, shows the SIP event call flow diagram for session establishment. Note that the session establishment is done between end-user and Media server via the B2BUA, although the content is streamed from a third party streaming server. Media server performs the media mixing and transcoding and
Figure 3: Call flow diagram for session establishment New IP
UE
S-CSCF One-Way RTP
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Media Server
Multimedia Source
One-Way RTP
Re-INVITE SDP4 Re-INVITE SDP4 200 OK SDP4 200 OK SDP4 One-Way RTP RTP Disconnect
Figure 4: Call flow diagram for Terminal Mobility Adaptation
• Stage 3: When the user switches terminal while the multimedia session is ongoing, the terminal issues a REFER request to the SIP A/S with the IP address of the new terminal and its capabilities. REFER request is forwarded to the media server which then encodes the media content to suit the new device capabilities such as the codec supported by the device and frame-rate the display supports and streams the multimedia content to the new device and disconnects RTP connection with the old terminal. Here again, negotiation and adaptation happens between the end user and the B2BUA-Media server combination and not with the streaming server, which would have then not only add delay to the negotiation process, but also may not have supported the codec used by the end-terminal to able to adapt the media content. New UE
UE
B2BUA S-CSCF One-Way RTP
Media Server
Multimedia Source One-Way RTP
REFER SDP5 REFER SDP5
200 OK SDP 6
200 OK SDP6
One-Way RTP RTP Disconnect
Figure 5: Call flow diagram for Session mobility Adaptation
6. Conclusion We have proposed a SIP-based, adaptive, qualityaware framework implementation for multimedia service delivery on IP Multimedia subsystem. The framework makes the following assumptions: 1. Terminal mobility occurs with in a single operator domain (Intra-operator mobility). 2. The multimedia stream is non-real-time in nature and the streaming server is SIPenabled. Although IETF specified SIP extensions make provision for supporting terminal and session mobility, the implementations have little scope for quality adaptation. In most cases, the session adaptation is performed by renegotiation between the source and receiver which are limited by their capabilities. Our contributions through this framework are: • Quality adaptation is performed by a mediator (SIP A/S and the media server combination) which is part of the operator network, hence it is highly controlled.
•
•
The Adaptation process is achieved by changing the encoding rate and format for the multimedia stream according to the changing receiver condition. The transcoding function of the media server facilitates quality adaptation. Hence adaptation is not limited by the source’s incapability of not supporting a certain encoding rate or format. Since media server supports most codec formats, renegotiation of codecs using SDP is straight forward and contributes less to the end-to-end delay. Unlike, the renegotiations between the source and receiver. When the source does not support the codec supported by the receiver, SDP renegotiations can get into a loop.
6.1. Future Work Experimental test bed preparation of the IMS architecture with the proposed implementation is in progress. Future work would involve evaluating the proposed mechanism on the test bed and quantify the performance in terms of delay incurred during the terminal and session mobility handoff, and due to the quality adaptation process. Also measure the userperceived QoS during and after mobility in terms of throughput obtained on the end-terminal. We also intend to make the quality adaptation process automated by sending wireless signal strength information via the SIP INFO method [23] to the SIP A/S and triggering the quality adaptation based on the signal strength on the receiver.
7. References [1]G. Camarillo, M.A. Garcia-Martin. The 3G IP Multimedia Subsystem. West Sussex: Wiley, 2004., [2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, and E. Schooler. Session Initiation Protocol (SIP) RFC 3261. IETF, A, 2002 [3] 3GPP TS 122.228: "IP Multimedia Subsystem (IMS); Stage 2, v.6.16.0,(Release 7) 2007. [4] 3GPP TS 122.228: "Service requirements for the Internet Protocol (IP) multimedia core network subsystem (IMS); Stage 1, v.7.5.0,(Release 7) 2006. [5] C. E. Perkins, IP Mobility Support for IPv4, RFC 3220,
Jan. 2002. [6] S. Das, A. McCauley, A. Dutta, A. Misra, K. Chakraborty and S. K. Das, “IDMP: An Intra-Domain Mobility Management Protocol for Next-Generation Wireless Networks”, IEEE Wireless Communications, Vol. 9, No. 3, Page(s): 38-45, June 2002. [7] A. C. Snoeren and H. Balakrishnan, “An End-to-End Approach to Host Mobility”, Proceedings MobiCom, Page(s): 155-166, August 2000
[8] F. Siddiqui and S. Zeadally, "Mobility Protocols for Handoff Management in Heterogeneous Networks", in Proceedings of 11th IFIP International Conference on Personal Wireless Communications (PWC'06), Lecture Notes in Computer Science (LNCS), Springer Verlag, Albacete, Spain, September 2006. [9] R. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T. Taylor, I. Rytina, M. Kalla, L. Zhang and V. Paxson, Stream control transmission protocol, RFC 2960 (2000). [10] H. Schulzrinne and E. Wedlund, "Application-Layer Mobility using SIP", ACM Mobile Computing and Communications Review, Vol. 4, No. 3, July 2000, pp. 47-57. [11] Paolo Bellavista, Antonio Corradi, Luca Foschini, "SIP-Based Proactive Handoff Management for Session Continuity in the Wireless Internet," icdcsw, p. 69, 26th IEEE International Conference on Distributed Computing Systems Workshops (ICDCSW'06), 2006. [12] F. Siddiqui and S. Zeadally, "Mobility Management across Hybrid Wireless Networks: Trends and Challenges", Computer Communications, Vol. 29, No. 9, Elsevier Science, 2006 [13] Wesley M. Eddy, "At What Layer Does Mobility Belong?", IEEE Communications Magazine, Vol. 42, Issue 10, pp. 155-159, October 2004. [14] R. Shacham et al., IETF draft-shacham-sippingsessionmobility- 02.txt, “Session Initiation Protocol (SIP) SessionMobility,” work in progress, Aug. 6, 2006 [15] Hand-off delay analysis in SIP-based mobility management in wireless networks ,Banerjee, N.; Basu, K.; Das, S.K. Parallel and Distributed Processing Symposium, 2003. Proceedings. International Volume , Issue , 22-26 April 2003 Page(s): 8 pp. [16] N. Nakajima, A. Dutta, S. Das and H. Schulzrinne,"Handoff delay analysis and measurement for SIP based mobility in IPv6," IEEE ICC, May 2003. [17] A. Johnston, S. Donovan, R. Sparks C. Cunningham, K. Summers Session Initiation Protocol (SIP) Basic Call Flow Examples RFC 3665. IETF, A, 2003 [18] R. Sparks, The Session Initiation Protocol (SIP) Refer Method, RFC 3515. IETF, A, 2003 [19] J. Van Dyke, E. Burger, A. Spitzer, Media Server Control Markup Language (MSCML) and Protocol RFC 4722. IETF, A, 2006 [20] M. Garcia-Martin, E. Henrikson, D. Mills, Private Header (PHeader) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP), RFC 3455. IETF, A., 2004 [21] M. Handley, V. Jacobson, C. Perkins, SDP: Session Description Protocol, RFC4566. IETF,A,2006. [22] E. Burger, J. Van Dyke, A. Spitzer, Basic Network Media Services with SIP, RFC 4240. IETF,A,2005 [23] S. Donovan, The SIP INFO Method, RFC 2976. IETF, A, 2000
