End-2-End QoS Provisioning in UMTS networks

Transcription

1 End-2-End QoS Provisioning in UMTS networks Haibo Wang Devendra Prasad October 28, 2004

2 Contents 1 QoS Support from end-to-end viewpoint UMTS IP Multimedia Subsystem (IMS) SIP protocol IMS Architecture End2End Service Architecture Before Session Setup Call Flows UMTS IP Aspects QoS Mapping between UMTS and IP or External Network Different Methods: IntServ and DiffServ Problems

3 Chapter 1 QoS Support from end-to-end viewpoint The goal of the project is to investigate the provisioning of end-to-end QoS for real-time applications using the PS domain of UMTS as access. This means the user satisfaction must be achieved on an application level rather than just on a network level. This chapter examines the E2E process of real-time/multimedia user sessions and identify the mechanisms relevant for QoS provisioning. 1.1 UMTS IP Multimedia Subsystem (IMS) The Release ąŕ99 architecture allows GSM/GPRS operators to gracefully evolve their networks to the UMTS architecture. However, deploying two separate networks for telephony and data services introduces severe limitations in terms of multimedia services and management of these networks. The UMTS Release 5 architecture proposes the addition of a new subsystem known as the IP Multimedia Subsystem (IMS) to the PS-CN for supporting traditional telephony as well as new multimedia services. IMS Telephony services PSTN/ISDN Voice over IP/ Multimedia sevices MS UTRAN PS CN Internet Internet Services Figure 1.1: High level Release 5 Architecture The IMS connects to both PSTN/ISDN as well as Internet. It may terminate voice and multi-media calls on both PSTN/ISDN and Internet. IMS not only supplies a convergence 3

4 CHAPTER 1. QOS SUPPORT FROM END-TO-END VIEWPOINT network solution for telephony and data services but also a higher level QoS management possibility for IP based real-time services, no matter the application servers locate in the IMS (the same operator s network) or on Internet SIP protocol Session Initiation protocol (SIP) has been chosen as the signaling and session control of packet telephony for establishing multimedia session in UMTS release 5 (R5) IMS. SIP is developed by the IETF. It is also implemented in 3GPP and 3GPP2 as the signaling protocol in the packet switch domain. SIP: SIP is used to establish, modify and terminate multimedia session over the external network or Internet. Multimedia session can be set of senders, receivers and the data streams flowing from the senders to the receivers. SIP is also used to invite third participant to an ongoing session, such as conferences. Through SIP session description, the participant can negotiate with the media types and other parameters of the session. SIP provides reliable transmission and run over different protocols such as TCP(), UDP(),and SCTP( Stream Control Transmission protocol) and also compatible with IPv4,IPv6. SIP is used for the managing multimedia communication due to following capabilities: Determination of destination userąŕs current location. Determination of user willingness to participate in a session. Determination of user terminal capabilities. Session Setup. Session Management Capabilities which includes modification of session parameters, invoking service function to provide service to a session and terminating a session. Since SIP is a client server protocol that uses a request and a response model. A SIP client means that any network element that has a capability to generate SIP request and receives a SIP response. On the other side, SIP server is a network element, which is used to service the SIP request and send back response to these elements. The SIP architecture contain four major components which is SIP User Agent (UA): A user agent is an external network endpoint (Internet) which is used to establish, modify and terminate session. A UA can act as a user agent client (UAC) and user agent server (UAS). UAC initiate SIP session request whereas UAS generate response to these request. SIP Redirect Server: A redirect server is also a UAS that is used to generate a response in order to redirect a request to other location. SIP Proxy Server: Proxy server can perform as UAC as well as UAS. It also act as an intermediary node between user and the destination. User SIP message is intercepted at SIP proxy server and SIP server route the message to the destination. 4

5 1.1. UMTS IP MULTIMEDIA SUBSYSTEM (IMS) SIP Registrar: The SIP registrar request is handled by UAS, which act as a registrar. It maintains the mapping from SIP user name to address and is the front end of the location service. The SIP has the following key aspects and will be explained through example. Naming and Addressing Message Location registration Session Establishment and Termination. Naming and Addressing: SIP user is uniquely identified by a SIP Uniform Resource Identifier (URI). SIP URI is similar to an address. For ex. Sip: hwano3@kom.auc.dk.where the Internet domain name of a SIP server in Haiboąŕs SIP service providerąŕs network is kom.auc.dk. SIPS URI is used in order to transport SIP message in a secure and encrypted way. Hence the format would be sips:hwan03@kom.auc.dk. In general, SIP URI format is : Sip: user: password@host: port; uri-parameters Headers. The user info contains username and password. If the user info is absent, then destination host acts as a resources and being identified. Otherwise the user field identifies a particular user at the host, and the password field specifies the associated password with the user. The user info is case sensitive and other fields are not. Message: Each SIP message can act as request or response message. The request message is sent from UAC to UAS whereas the response message is sent from UAS to UAC. Since, SIP is a text-based protocol. Every SIP message consists of the following: A Start-line One or more header field An empty line indicating the end of the header field. An optional message body. Example: Start line INVITE sip: hwan03@kom.auc.dk SIP/2.0 5

6 CHAPTER 1. QOS SUPPORT FROM END-TO-END VIEWPOINT Header Field Empty Line Via: SIP/2.0/UDP aau.edu.dk:5060;branch=z1hgw4b Max-forwards:70 To: haibo< From:Dev< >;tag= Call-ID: aau.edu.dk Cseq:123456INVITE Cantact:< Dev > Content-type:application/sdp Content Length:32 Message body v=0 1. The Start line indicate the SIP INVITE request usedto invite user to a SIP session, Where SIP/2.0 is the SIP protocol version name. The status ĺcline of the response message may look like SIP/ The header field used to carry information needed to route a request or to manage a SIP session. The Header Fields contains the following information: To: Specified the desired recipient of the SIP session request. From: Specifies the initiator of the request. Subject: Specifies the subject of the session. Call-ID : Contain a unique identifier of the session. It is used to identify message that belong to the same call or session. Via: Indicates the transport-layer protocol used for the transaction and identifies the location where the response message for this request should be forwarded. Contact: Contains a SIP URI that can be used for requests. On comparing with the Via header field, it indicates the forwarding path for the response message. Also acts as a redirect server. Max Forward: indicates the maximum number of hops a request can traverse on its way to the destination. Cseq: Contain a sequence number and a method name. It is incremented with each new request. Content-Type: indicates that the message body is a description of the session in the format of the Session Description protocol. Location Registration: SIP provides a location service with the help of UAS (SIP Registrar) residing in a userąŕs SIP service provider network. It is responsible for maintaining userąŕs current location 6

7 1.1. UMTS IP MULTIMEDIA SUBSYSTEM (IMS) Example 2a: The UA Side Start line REGISTER sip: SIP/2.0 Header Field Via: SIP/2.0/UDP hwan03-pc.aau.edu.dk: 5060;branch=z1hgw4b Max-forwards:70 To: haibo < From: Dev<sip: >;tag= Call-ID: aau.edu.dk Cseq: 2660REGISTER Contact: < Dev > Content-type: application/sdp Content Length:0 Expires: 3600 Example 2a shows the registration process originating from UA. The UA of the user Hwan03 send a Registration request to a registrar. The registrar can be pre-configured. If it is not pre-configured then the request can be sent to the host part of the address-of ĺcrecord (sip: kom.auc.dk). The UA may also multicast to all SIP servers. The Header field consists of few new header compared to the previous example. Expire 3600: Indicates that registration is valid for only 3600 seconds Content Length : It indicates that there is no message body as the length is 0. After receiving the registration request, the Registrar process the request as shown in Example 2b. Example 2a:The Registrar Side Start line SIP/ OK Header Field Via: SIP/2.0/UDP hwan03-pc.aau.edu.dk:5060;branch=z1hgw4b, received = Max-forwards:70 To: haibo < sip:hwan03@kom.auc.dk> From: Dev<sip: hwan03@kom.auc.dk >;tag= Call-ID: f25dbj567js@ aau.edu.dk Cseq: 2660REGISTER Contact: < Dev dpra03@ > Content-type: application/sdp Content Length:0 Expires: 3600 The registrar may authenticate the user and then can decide about the user authorization to change the registration record. The user gets 403 Forbidden in the case of failed authorization and authentication. If userąŕs binding updates are successfully completed then the user gets the 200 OK messages as shown in Figure 1. Session Establishment and Termination The establishment of the SIP session can be done through peer-to-peer mode or a server 7

8 CHAPTER 1. QOS SUPPORT FROM END-TO-END VIEWPOINT mode. When a Caller establishes a call to a callee directly without going through the SIP server is known as Peer-to-Peer mode. In this scenario, the caller must know the callee current location. In the SIP Server mode, a caller send its SIP request message to a SIP server in the callerąŕs or callee ąős SIP service provider network. So, itąŕs a SIP server, which helps the caller in establishing the session with the callee after locating their current location. This SIP server is also called as Proxy Server.The Proxy server may rewrite the message before forwarding it to the destination depending upon the network conditions. A SIP server may not relay the message forward to the destination instead giving a path indication to the caller about the next contact point to reach the callee. This type of SIP Server is called as Redirect Server IMS Architecture Now letąŕs introduce the components of the IP Multimedia Subsystem. The first key elements are the Call Session Control Functions (CSCF) and the Home Subscriber Server. The CSCF has taken the majority of the MSC functionality in the IMS architecture. The CSCF is analogous to the SIP server in the IETF architecture.?? Figure 1.2: IP Multimedia Subsystem Call Session Control Functions (CSCF) The first component that needs to be discussed is the Call Session Control Function (CSCF.) Its function is to process signaling messages for controlling the userąŕs multimedia session. 8

9 1.2. END2END SERVICE ARCHITECTURE The existing packet switched core network is used to support the bearer path for the multimedia session and the CSCFs are used to establish the sessions and perform features. The CSCFs perform a number of functions. The first is the multimedia session control function. This is an evolution of the MSC call control function. Next is the address translation function (i.e. evolution of the digit translation function.) The CSCF must also perform services switching for services and vocoder negotiation. The CSCF must perform the handling of the subscriber profile (i.e. the VLR.) The CSCF can play three roles: the Proxy CSCF (P-CSCF) role, the Interrogating CSCF (I-CSCF) role and the Serving CSCF (S-CSCF) role. The P-CSCF is the mobiles first point of contact in the IMS network. The I-CSCFąŕs function is to determine the S-CSCF based on load or capability. The S-CSCF is responsible for the mobileąŕs session management. All three of these roles can support the firewall capability. To focus on QoS related issues, we will not discuss the case that the user is roaming in a visited network hence only S-CSCF need to be explained in detail. Serving CSCF ĺc (S-CSCF) The Serving Call Session Control Function (S-CSCF) is the node that performs the session management for the IMS network. There can be several S-CSCFs in the network. They can be added as needed based on the capabilities of the nodes or the capacity requirements of the network. The S-CSCF in the home network is responsible for all session control. Home Subscriber Server As in the legacy mobile network, there is still a need for a centralized subscriber database. The Home Location Register (HLR) has evolved into the Home Subscriber Server (HSS.) The HSS interfaces with the I-CSCF and the S-CSCF to provide information about the location of the subscriber and the subscriberąŕs subscription information. Media Gateway (MGW) is the brawn. It is the workhorse that does the processing of the media bits between end users. Its primary function is to convert media from one format to another. In UMTS this will predominantly be between Pulse Code Modulation (PCM) in the PSTN and an IP based vocoder format. Media Gateway and Media Gateway Control Function The IMS supports several nodes for interworking with PSTN networks. These are the Media Gateway (MGW), the Media Gateway Control Function (MGCF), and the Transport Signaling Gateway (TSGW). They are not relevant to this project hence will not be mentioned further more. 1.2 End2End Service Architecture This section examines an E2E real-time service process, assuming a MS at its home network requires an application from an application server, and the P-CSCF, I-CSCF and 9

10 CHAPTER 1. QOS SUPPORT FROM END-TO-END VIEWPOINT S-CSCF locate in the same network node. This case study is based on the UMTS R5 IMS structure and SIP is the application level En2End signalling connecting the whole process. The whole process and all network elements involved are depicted in figure 1.3. The IMS services architecture allows deployment of new services by operators and 3rd party service providers. This provides subscribes a wide choice of services.the IMS defines three different was of delivering services. These are explained below: Native SIP Services:One or more SIP application servers may be used to deploy services inside the operator s IMS networks. 3rd party services:umts has defined Open Services Access (OSA) to allow 3rd party service providers to offer services through UMTS network. The OSA offers a secure API for 3rd party service providers to access UMTS networks. Therefore, subscribes are not restricted to the services offered by the operators. Intelligent Network(IN) services: While new and innovative services are required, the telephony services cannot be ignored. The release ąŕ99 networks use CAMEL (Customized Applications for Mobile Enhanced Logic) Service Environment for deploying intelligent networking services such as pre-paid service and 1-800service. These application can still be supported in IMS via certain interface function between SIP and CAMEL. Figure 1.3: End2End Real-time Service Architecture in 3GPP 10

11 1.2. END2END SERVICE ARCHITECTURE Before Session Setup 1. System acquisition: The first step is to power on the mobile and lock on to the UMTS system. Once the appropriate cell is selected, the UMTS mobile is ready to communicate signaling messages required to establish a data session. 2. Data Connection Setup: Once the system has been acquired, the next step is to establish the data connection or ąřpipeąś to the SIP and other services. The UE does not know the IP address of the Proxy-CSCF at this point to perform a SIP registration. The data connection is completed in a two-step process using Attach and Packet Data Protocol (PDP) Context Activation message sequences, as described in chapter 2. This establishes the path required to carry SIP related signaling messages to the Proxy-CSCF through the GGSN, which is the gateway to the Proxy-CSCF. Therefore, the Attach and the PDP Context Activation are two key steps required to create a data ąřpipeąś to the Proxy CSCF for SIP services. The response to the PDP Context Activation message also includes the identity of the Proxy-CSCF for the UE to use to perform the registration process. 3. Service Registration: Before establishing an IP Multimedia session, the UE needs to perform the Service registration operation to let the IMS network know the location of the UE. This registration is an application or SIP registration for various SIP services. The UMTS UE acts as a SIP client and sends a SIP registration message to its home system through the Proxy-CSCF. 4. Session Setup: After a PDP context is activated and Service Registration is finished, the UE can establish a session Call Flows Figure 1.3 shows the flow of signaling messages and the flow of the user data. 1st signaling messages, the signaling messages will go from the mobile through the UTRAN, to the SGSN/GGSN, out to the CSCFąŕs and out to the application servers (either to an internal SIP supported server, or out to an external 3nd-party server which support SIP, where the operator could even supply a SIP server in its network with an open interface to 3nd-party application servers located on Internet.) It is important to note which of these components are processing the message versus the components that are ąřroutingąś the message. At the time the mobile is sending a request to establish a service, this request is sent to S-CSCF (via the Proxy and the Interrogating) to request the service. The SGSN and the GGSN will only perform the function of routers. They do not look at the contents of the message, they only look at the destination IP address and route the message accordingly. 2nd user data As can be seen from the figure, the user information will flow from the mobile, through the SGSN and the GGSN out to the application server. It will bypass the CSCF all together. This is important to note that SIP is an off-band signalling like SS7 in ISDN, which take different paths with user data through the network. 11

12 CHAPTER 1. QOS SUPPORT FROM END-TO-END VIEWPOINT 1.3 UMTS 1.4 IP Aspects 1.5 QoS Mapping between UMTS and IP or External Network Different Methods: IntServ and DiffServ Problems 12