KR20070001559A - Method of guarantee for end-to-end quality of service, using sip and rsvp-te - Google Patents
Method of guarantee for end-to-end quality of service, using sip and rsvp-te Download PDFInfo
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- H—ELECTRICITY
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- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1069—Session establishment or de-establishment
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/18—End to end
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/1045—Proxies, e.g. for session initiation protocol [SIP]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/1046—Call controllers; Call servers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
Abstract
Description
도1은 시스템 아키텍처1 is a system architecture
도2는 SIP/SDP, RSVP-TE를 연동한 자원 할당 절차2 is a resource allocation procedure interworking with SIP / SDP and RSVP-TE
도3은 본 발명에서 제안하는 단말, Service Provider간의 구성모듈3 is a configuration module between a terminal and a service provider proposed in the present invention
도4는 Activity Diagram4 is Activity Diagram
PDA, Host PC와 같은 단말간의 IP Data communication에 있어서 QoS(Quality of Service)가 보장된 서비스를 제공하기 위함이다. 기존에 IntServ등을 이용하여 End user간의 QoS 보장 제공 시도 하였지만, Complexity 문제로 인해 대규모 시스템에 적용하기에는 불가능 하고, 이를 극복하고자 User Traffic를 Class별로 aggregation시키는 Diffserv 기술이 등장 하였으나 이 역시 Core Network에서는 실 효성을 거둘 수 있으나, End-User단까지의 QoS보장에는 한계가 있다. This is to provide guaranteed service (QoS) in IP data communication between terminals such as PDAs and host PCs. In the past, attempts were made to provide QoS guarantees between end users using IntServ. However, due to the complexity problem, it is impossible to apply it to large-scale systems. To overcome this problem, Diffserv technology that aggregates user traffic by class has emerged. However, there is a limit to QoS guarantee to end-user level.
본 발명에서는 기존의 SIP(Session Initiation Protocol)/SDP(Session Description Protocol)를 이용하여 단말간의 Session를 설정하고, Session 설정 시 단말용 RSVP-TE agent를 이용, ISP의 RSVP-TE Server/CAC(Call Admission Control)과 UNI (User-Network Interface) Signaling을 통하여 QoS 보장형 Packet flow를 설정함으로써 End-To-End간 QoS를 보장해 준다.In the present invention, a session between terminals is established using an existing Session Initiation Protocol (SIP) / Session Description Protocol (SDP), and an RSVP-TE Server / CAC (Call) of an ISP is used by using an RSVP-TE agent for a terminal when establishing a session. QoS is guaranteed between end-to-end by setting QoS guaranteed packet flow through Admission Control) and UNI (User-Network Interface) signaling.
본 발명은 IP Data communication에 있어서 단말간의 End-To-End QoS(Quality of Service)를 제공하기 위한 방안으로서 아래와 같이 4 Part로 구분된다. The present invention is to provide end-to-end quality of service (QoS) between terminals in IP data communication. The present invention is divided into 4 parts as follows.
-MMCoordinator-MMCoordinator
-SIP ProxyServer, Registration ServerSIP ProxyServer, Registration Server
-RSVP-TE(Resource Reservation Protocol-Traffic engineering) Server/CAC(Call Admission Control) -RSVP-TE (Resource Reservation Protocol-Traffic engineering) Server / CAC (Call Admission Control)
-NMS(Network Management System)NMS (Network Management System)
위의 4 part의 전체적인 Architecture는 도1과 같이세부적으로 아래와 같은 구성 모듈을 가진다. The overall architecture of the above 4 parts has the following configuration modules in detail as shown in FIG.
1.MMService Coordinator1.MMService Coordinator
1.1.SIP/SDP1.1.SIP / SDP
1.2.RSVP-TE Agent1.2.RSVP-TE Agent
2.SIP ProxyServer/ Registration Server2.SIP ProxyServer / Registration Server
3.RSVP-TE Server/CAC(Call Admission Control)3.RSVP-TE Server / CAC (Call Admission Control)
3.1.SLA(Service Level Agreement) List3.1.SLA (Service Level Agreement) List
4.NMS(Network Management System)4.NMS (Network Management System)
4.1.Diffserv-over-MPLS4.1.Diffserv-over-MPLS
4.2.Network Information List 4.2.Network Information List
MMService Coordinator에서는 SIP/SDP, RSVP-TE agent 모듈로 나뉘어 진다. SIP/SDP는 단말간의 QoS Guaranteed Session에 대한 연결 설정을 관리하며, SIP를 이용하여 기본 적인 session설정을 하고, SDP를 이용하여 추가적으로 요구되는 QoS 속성을 전달한다. SIP/SDP를 이용하여 단말간의 QoS Guaranteed session 설정 시, 자원을 할당하기 위한 protocol로 RSVP-TE가 사용된다. RSVP-TE agent는 SIP/SDP로부터 QoS 속성에 대한 정보를 획득한 후, Service Provider의 RSVP-TE Server에게 자원 할당하는 기능을 가진다.MMService Coordinator is divided into SIP / SDP, RSVP-TE agent module. SIP / SDP manages connection setting for QoS Guaranteed Session between terminals, sets basic session using SIP, and delivers additionally required QoS attributes using SDP. When establishing a QoS guaranteed session between terminals using SIP / SDP, RSVP-TE is used as a protocol for allocating resources. The RSVP-TE agent has a function of allocating resources to the RSVP-TE Server of the Service Provider after obtaining information on QoS attributes from the SIP / SDP.
SIP ProxyServer/Registration Server를 살펴보면 ProxyServer는 Terminal, SIP Server들 간의 SIP(Session Initiation Protocol) message 처리를 담당한다. Registration Server에서는 사용자들에 대한 정보들을 관리하는 기능을 담당한다. Looking at SIP ProxyServer / Registration Server, ProxyServer is in charge of handling Session Initiation Protocol (SIP) messages between Terminal and SIP Servers. Registration Server is responsible for managing information about users.
RSVP-TE Server/CAC(Call Admission Control)를 살펴보면, RSVP-TE Server는 단말의 RSVP-TE Agent로부터 요청되는 자원 할당 요청을 처리 한다. RSVP-TE Server가 송신단 RSVP-TE agent로부터 자원 할당 요청을 받게 되면, 송수신자의 IP Address와 TCP/UDP Port번호를 CAC로 넘겨준다. CAC로부터 자원 요청에 대한 허가가 떨어지면 RSVP-TE Server는 수신단의 RSVP-TE agent로 자원 요청 signaling을 포워딩하게 되고, 수신 단에서 자원 요청을 받아들이게 되면 지속적으로 설정된 자원을 관리한다. CAC에서는 RSVP-TE Server로부터 요청된 자원 할당에 관해서 검토한다. Looking at RSVP-TE Server / CAC (Call Admission Control), RSVP-TE Server processes the resource allocation request requested from the RSVP-TE Agent of the terminal. When the RSVP-TE Server receives a resource allocation request from the sending RSVP-TE agent, it transmits the IP address and TCP / UDP port number of the transceiver to the CAC. When the permission for the resource request is dropped from the CAC, the RSVP-TE server forwards the resource request signaling to the RSVP-TE agent of the receiving end. When the receiving end accepts the resource request, the RSVP-TE server manages the resource continuously configured. The CAC reviews the resource allocation requested from the RSVP-TE Server.
이때 SLA(Service Level Agreement) List를 이용하여 Service-Provider와 Service-User간의 사전에 계약된 내용들을 검토, 계약 내용을 준수 할 경우 NMS(Network Management System)에게 자원 요청을 한다.In this case, SLA (Service Level Agreement) List is used to review the contents of the contract between Service-Provider and Service-User, and if the contract is followed, resource request is made to NMS (Network Management System).
NMS(Network Management System)은 Diffserv-over-MPLS 기능을 이용하여 End User간의 QoS 서비스를 제공하며, 추가적으로 QoS요청 단말간의 Network 도달 정보 List를 유지한다. Network Management System (NMS) provides QoS service between end users by using Diffserv-over-MPLS function, and additionally maintains network reach information list between QoS request terminals.
SIP/SDP, RSVP-TE를 연동한 자원 할당 절차는 도2와 같다.A resource allocation procedure interworking with SIP / SDP and RSVP-TE is shown in FIG.
User Agent A가 INVITE를 Message를 이용하여 User Agent B를 초대 한다 여기서 SIP ProxyServer들은 SIP message를 포워딩 하는 역할을 제공한다. User Agent A가 INVITE Message를 작성할 때 QoS에 관한 정보를 SDP에 기록한다. INVITE message를 받은 User Agent B는 SDP의 정보를 분석하고 QoS의 요청이 있는 것을 확 인하고 183 Session Progress Message를 User Agent A에게 보낸다. User Agent A invites User Agent B using INVITE as a Message. SIP ProxyServers provide the role of forwarding SIP messages. When User Agent A creates an INVITE message, it records information about QoS in SDP. Receiving the INVITE message, User Agent B analyzes the SDP information, confirms that there is a request for QoS, and sends a 183 Session Progress Message to User Agent A.
183 Message를 받은 User Agent A는 User Agent B가 QoS 요청에 대해 수락한 것에 대한 응답으로 Prack message를 보낸다. User Agent A receives the 183 message and sends a Prack message in response to User Agent B accepting the QoS request.
Prack message를 받은 User Agent B는 RSVP-TE agent를 통하여 User Agent B -> User Agent A로 단 방향 Path를 설정하게 되고, Prack message에 대한 응답 message로 200 OK message를 User Agent A에게 보낸다. 200 OK 메시지를 받는 User Agent A는 User Agent B가 User Agent B -> User Agent A 간 Path간 설정 된 것을 알고 User Agent A -> User Agent B가 단 방향 Path를 설정한다. User Agent A가 Path를 설정하고 난 후, 실질적으로 두 단말간에 Media stream을 송수신하기 위해 Update message를 User Agent B에게 보낸다. The user agent B receiving the prack message sets the one-way path from the user agent B-> user agent A through the RSVP-TE agent, and sends a 200 OK message to the user agent A as a response message to the prack message. User Agent A, which receives 200 OK message, knows that User Agent B is set between the path between User Agent B and User Agent A. Then, User Agent A and User Agent B set one-way path. After User Agent A sets the path, it actually sends an Update message to User Agent B to send and receive media streams between the two terminals.
User Agent B는 Update message를 확인하고 이에 대한 응답으로 200 OK를 User Agent A로 보내게 되고, 이를 받은 User Agent는 INVITE에 관한 session negotiation를 마치고 실질적으로 QoS가 보장된 data communication을 하게 된다.User Agent B checks the update message and sends 200 OK to User Agent A in response. The received User Agent completes the session negotiation on INVITE and performs data communication with guaranteed QoS.
도3은 본 발명에서 제안하는 단말, Service Provider간의 구성모듈이다. 도3에 의하면 단말에서 SIP/SDP, RSVP-TE Agent기능을 제공하고, Service Provider에서는 하나의 Router 또는 스위치에 위의 모든 기능들을 구성 할 수도 있고 이를 부분적으로 나누어 분산처리 시켜 구성 할 수도 있다. 3 is a configuration module between a terminal and a service provider proposed in the present invention. According to FIG. 3, the terminal provides SIP / SDP and RSVP-TE Agent functions, and the Service Provider may configure all the above functions in one router or switch, or may divide and partially divide the above functions.
도4의 Activity Diagram은 RSVP-TE agent와 RSVP-TE Server/CAC, NMS 상호연동부분을 보여준다.4 shows the RSVP-TE agent, RSVP-TE Server / CAC, and NMS interworking parts.
I. IntroductionI. Introduction
Traffic usage has been growing at an exponential rate. The DiffServ model was proposed as a solution to the scalability problem of the IntServ model. In this case, packets are classified into a small number of service classes at the edge router according to their service requirements. The core router will differentiate between packets on a class-by-class rather than flow-by-flow. It avoids the scalability problem associated with IntServ.Traffic usage has been growing at an exponential rate. The DiffServ model was proposed as a solution to the scalability problem of the IntServ model. In this case, packets are classified into a small number of service classes at the edge router according to their service requirements. The core router will differentiate between packets on a class-by-class rather than flow-by-flow. It avoids the scalability problem associated with IntServ.
At the same time, the Session Initiation Protocol (SIP) and Recourse Reservation Protocol (RSVP) have become the key components to provide the required QoS. In order to provide QoS-guaranteed realtime multimedia services on IP-based network: (i) end-to-end signaling to establish a session, and (ii) UNI/NNI signaling to establish QoS and bandwidth-guaranteed scheme for data flow. For the first purpose, SIP/SDP (Session Description Protocol) can be deployed, while RSVP-TE will be used to organize the UNI and NNI signaling to establish QoS-guaranteed connection.At the same time, the Session Initiation Protocol (SIP) and Recourse Reservation Protocol (RSVP) have become the key components to provide the required QoS. In order to provide QoS-guaranteed realtime multimedia services on IP-based network: (i) end-to-end signaling to establish a session, and (ii) UNI / NNI signaling to establish QoS and bandwidth-guaranteed scheme for data flow. For the first purpose, SIP / SDP (Session Description Protocol) can be deployed, while RSVP-TE will be used to organize the UNI and NNI signaling to establish QoS-guaranteed connection.
The Session Initiation Protocol (SIP) is a new signaling, presence and instant messaging protocol developed to set up, modify, and tear down multimedia sessions, request and deliver presence and instant messages[1]. In order to establish multimedia communication session, SIP/SDP messages must be exchanged between session participants to check out the participant capability, media type negotiation and format type.The Session Initiation Protocol (SIP) is a new signaling, presence and instant messaging protocol developed to set up, modify, and tear down multimedia sessions, request and deliver presence and instant messages [1]. In order to establish multimedia communication session, SIP / SDP messages must be exchanged between session participants to check out the participant capability, media type negotiation and format type.
After negotiation of session parameters, QoS-quaranteed connection for the present session should be established between session participants. For that purpose RSVP-TE (Resource Reservation Protocol with extension of Traffic Engineering) will be used.After negotiation of session parameters, QoS-quaranteed connection for the present session should be established between session participants. For that purpose RSVP-TE (Resource Reservation Protocol with extension of Traffic Engineering) will be used.
In this paper we propose a new functional model for QoS-guaranteed DiffServ Provisioning. Based on the proposed architecture, QoS-guaranteed DiffServ Provisioning can be efficiently implemented and deployed on the IP-based networks of different operators. We also analyze the issues related to the interaction between SIP/SDP and RSVP-TE protocols in the new developed model called Coordinator, focusing on the main features of the mentioned protocols.In this paper we propose a new functional model for QoS-guaranteed DiffServ Provisioning. Based on the proposed architecture, QoS-guaranteed DiffServ Provisioning can be efficiently implemented and deployed on the IP-based networks of different operators. We also analyze the issues related to the interaction between SIP / SDP and RSVP-TE protocols in the new developed model called Coordinator, focusing on the main features of the mentioned protocols.
The rest of this paper is organized as the follows. In section II, related works are briefly described. In section III, system architecture will be explained. Section IV is dedicated to implementation and analysis of proposed model. Finally, we will conclude this paper in section V.The rest of this paper is organized as the follows. In section II, related works are briefly described. In section III, system architecture will be explained. Section IV is dedicated to implementation and analysis of proposed model. Finally, we will conclude this paper in section V.
II. Related WorksII. Related Works
2.1 Session Initiation Protocol (SIP) and Session Description Protocol (SDP)2.1 Session Initiation Protocol (SIP) and Session Description Protocol (SDP)
SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility - users can maintain a single externally visible identifier regardless of their network location [2].SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility-users can maintain a single externally visible identifier regardless of their network location [2].
SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build complete multimedia architecture. SIP does not provide services. Rather, SIP provides primitives that can be used to implement different services. For instance, SIP can locate a user and deliver an opaque object to his current location. If this primitive is used to deliver a session description written in SDP, for instance, the endpoints can agree on the parameters of a session.SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build complete multimedia architecture. SIP does not provide services. Rather, SIP provides primitives that can be used to implement different services. For instance, SIP can locate a user and deliver an opaque object to his current location. If this primitive is used to deliver a session description written in SDP, for instance, the endpoints can agree on the parameters of a session.
SIP does not offer conference control services such as flow control or voting and does not prescribe how a conference is to be managed. SIP can be used to initiate a session that uses some other conference control protocol. Since SIP messages and the sessions they establish can pass through entirely different networks, SIP cannot, and does not, provide any kind of network resource reservation capabilities.SDP was proposed to convey information about media streams in multimedia sessions to permit the recipients of a session description to take part in the session [3]. Session description information, such as information about the bandwidth to be used by the session, type of media, time(s) the session is active, transport protocol (RTP/UDP/IP, H.320, etc), the format of the media (H.261 video, MPEG video, etc) can be specified based on the SDP.SIP does not offer conference control services such as flow control or voting and does not prescribe how a conference is to be managed. SIP can be used to initiate a session that uses some other conference control protocol. Since SIP messages and the sessions they establish can pass through entirely different networks, SIP cannot, and does not, provide any kind of network resource reservation capabilities.SDP was proposed to convey information about media streams in multimedia sessions to permit the recipients of a session description to take part in the session [3]. Session description information, such as information about the bandwidth to be used by the session, type of media, time (s) the session is active, transport protocol (RTP / UDP / IP, H.320, etc), the format of the media (H.261 video, MPEG video, etc) can be specified based on the SDP.
Since multimedia communication session will be fully determined by SIP/SDP, based on the required connection parameters connection establishment will be claimed by UNI signaling function.Since multimedia communication session will be fully determined by SIP / SDP, based on the required connection parameters connection establishment will be claimed by UNI signaling function.
2.2 RSVP-TE2.2 RSVP-TE
The RSVP protocol is used by a host to request specific qualities of service from the network for particular application data streams or flows. RSVP is also used by routers to deliver quality-of-service(QoS) requests to all nodes along the path(s) of the flows and to establish and maintain state to provide the requested service. RSVP requests will generally result in resources being reserved in each node along the data path[4].The RSVP protocol is used by a host to request specific qualities of service from the network for particular application data streams or flows. RSVP is also used by routers to deliver quality-of-service (QoS) requests to all nodes along the path (s) of the flows and to establish and maintain state to provide the requested service. RSVP requests will generally result in resources being reserved in each node along the data path [4].
As the name suggests, the RSVP extensions are primarily intended for traffic engineering. In order to support traffic engineering, RSVP-TE makes a few additions to traditional RSVP. First, it expands the definition of traffic flows to include label switched paths. That lets and RSVP-capable router assign traffic to RSVP flow based on their MPLS label. Second, RSVP-TE modifies the protocol rules so path and reservation messages can travel between ingress and egress routers. In the RSVP-TE messages themselves, RSVP-TE adds fields that let path messages request label mappings and reservation messages return label values. The RSVP-TE extensions also allow an ingress router to specify an explicit route for the flow, much the same way as CR-LDP. Finally, RSVP-TE makes the actual resource reservation optional. With this flexibility, routers can use RSVP-TE to distribute labels even when the paths do not need reserved network resources[5].As the name suggests, the RSVP extensions are primarily intended for traffic engineering. In order to support traffic engineering, RSVP-TE makes a few additions to traditional RSVP. First, it expands the definition of traffic flows to include label switched paths. That lets and RSVP-capable router assign traffic to RSVP flow based on their MPLS label. Second, RSVP-TE modifies the protocol rules so path and reservation messages can travel between ingress and egress routers. In the RSVP-TE messages themselves, RSVP-TE adds fields that let path messages request label mappings and reservation messages return label values. The RSVP-TE extensions also allow an ingress router to specify an explicit route for the flow, much the same way as CR-LDP. Finally, RSVP-TE makes the actual resource reservation optional. With this flexibility, routers can use RSVP-TE to distribute labels even when the paths do not need reserved network resources [5].
After determination of QoS & traffic parameters for a multimedia session, QoS guaranteed per-class-type connections for the session must be establish among the participant terminals. In IP/MPLS network, connection establishment is accomplished by UNI(user-interface-interface) and NNI(network node interface)signaling. For UNI signaling between user terminal and ingress edge router, RSVP-TE can be used to carry the connection request. In order to support per-class-type DiffServ provisioning, RSVP-TE must provide traffic engineering extensions so as to deliver the traffic & QoS parametersAfter determination of QoS & traffic parameters for a multimedia session, QoS guaranteed per-class-type connections for the session must be establish among the participant terminals. In IP / MPLS network, connection establishment is accomplished by UNI (user-interface-interface) and NNI (network node interface) signaling. For UNI signaling between user terminal and ingress edge router, RSVP-TE can be used to carry the connection request. In order to support per-class-type DiffServ provisioning, RSVP-TE must provide traffic engineering extensions so as to deliver the traffic & QoS parameters
The multi-media coordinator must provide RSVP-TE client function, while the ingress edge IP/MPLS router must support the RSVP-TE server function. Since RSVP-TE establishes only unidirectional connection, two PATH-RESV message exchanges should be implemented to establish bi-directional path between user terminal and ingress router[6].The multi-media coordinator must provide RSVP-TE client function, while the ingress edge IP / MPLS router must support the RSVP-TE server function. Since RSVP-TE establishes only unidirectional connection, two PATH-RESV message exchanges should be implemented to establish bi-directional path between user terminal and ingress router [6].
III. Architecture for SIP and RSVP-TE for QoS-guaranteed DiffservIII. Architecture for SIP and RSVP-TE for QoS-guaranteed Diffserv
3.1 SIP with RSVP-TE architecture3.1 SIP with RSVP-TE architecture
This article presents QoS guaranteed realtime multimedia service architecture. (fig. 5) shows the proposed architecture to provide QoS guaranteed realtime multimedia service .This article presents QoS guaranteed realtime multimedia service architecture. (fig. 5) shows the proposed architecture to provide QoS guaranteed realtime multimedia service.
(fig. 5) Architecture for SIP with RSVP-TE(fig. 5) Architecture for SIP with RSVP-TE
This architecture consists of three main modules : (ⅰ) User Agent , (ii) SIP proxy server, and (ⅲ) RSVP-TE server. User agent consists of : (ⅰ) coordinator, (ⅱ) SIP, (iii) audio/video connections, and (Ⅳ) RSVP-TE. The coordinator manages these 3 modules to function systematically.This architecture consists of three main modules: (iii) User Agent, (ii) SIP proxy server, and (iii) RSVP-TE server. User agent consists of: (i) coordinator, (ii) SIP, (iii) audio / video connections, and (IV) RSVP-TE. The coordinator manages these 3 modules to function systematically.
The SIP proxy server forwards message between user agents and takes care of user agent registration. The SIP module in the service requestor first sends INVITE message to the service recipient. In SIP module, the required bandwidth is usually sent SDP message during INVITE request.The SIP proxy server forwards message between user agents and takes care of user agent registration. The SIP module in the service requestor first sends INVITE message to the service recipient. In SIP module, the required bandwidth is usually sent SDP message during INVITE request.
For the resource allocation, the RSVP-TE user agent sends Path message to the RSVP server at NMS. The RSVP-TE server calls NMS's Call Admission control (CAC) to check the availability of the resources. The CAC checks with the available resources and notifies RSVP-Server. When requested resource is unavailable,For the resource allocation, the RSVP-TE user agent sends Path message to the RSVP server at NMS. The RSVP-TE server calls NMS's Call Admission control (CAC) to check the availability of the resources. The CAC checks with the available resources and notifies RSVP-Server. When requested resource is unavailable,
RSVP server notifies error message to the requestor. If resource is available, RSVP-server forwards the Path message to the recipient User agent. The user agent responds with RESV message to the RSVP server, which will make the resource allocated.RSVP server notifies error message to the requestor. If resource is available, RSVP-server forwards the Path message to the recipient User agent. The user agent responds with RESV message to the RSVP server, which will make the resource allocated.
3.2 Signaling mechanism between SIP and RSVP-TE3.2 signaling mechanism between SIP and RSVP-TE
The basic idea is to let the terminals start a RSVP based bandwidth reservation during the SIP session setup. The SIP user agents, that should include the RSVP-TE support, are connected to an DiffServ based IP network; all routers are DiffServ aware and support RSVP-TE, per flow bandwidth reservation and per flow packet scheduling. According to this scenario, when the calling User Agent wants to establish a QoS call, it sends the SIP INVITE message to the callee, specifying that a bandwidth reservation is requested. Upon the receiving of the INVITE message, a 183 Session Progress response is sent from the callee and then the resource reservation procedure can start. Depending if the bandwidth reservation is requested for one or two-ways traffic flows, the caller or/and the callee starts a RSVP session by sending PATH messages to the peer party, followed by the RSVP signaling flow. Upon the reception of the RESV messages each user agent realizes that the reservation has been successfully setup and the SIP session setup can continue with the 180 Ringing message, the 200 OK and the caller ACK.The basic idea is to let the terminals start a RSVP based bandwidth reservation during the SIP session setup. The SIP user agents, that should include the RSVP-TE support, are connected to an DiffServ based IP network; all routers are DiffServ aware and support RSVP-TE, per flow bandwidth reservation and per flow packet scheduling. According to this scenario, when the calling User Agent wants to establish a QoS call, it sends the SIP INVITE message to the callee, specifying that a bandwidth reservation is requested. Upon the receiving of the INVITE message, a 183 Session Progress response is sent from the callee and then the resource reservation procedure can start. Depending if the bandwidth reservation is requested for one or two-ways traffic flows, the caller or / and the callee starts a RSVP session by sending PATH messages to the peer party, followed by the RSVP signaling flow. Upon the reception of the RESV messages each user agent realizes that the reservation has been successfully setup and the SIP session setup can continue with the 180 Ringing message, the 200 OK and the caller ACK.
(fig. 6) shows the SIP/RSVP-TE signaling flow for the call setup between two SIP/RSVP-TE aware user agents. Figure . The SIP/RSVP call setup signaling flow.(fig. 6) shows the SIP / RSVP-TE signaling flow for the call setup between two SIP / RSVP-TE aware user agents. Figure. The SIP / RSVP call setup signaling flow.
(fig. 6) QoS setup with SIP messages(fig. 6) QoS setup with SIP messages
On the figure above, User Agent (UA) A sends INVITE message includes quality of service preconditions in the SDP. After receiving 183 Session Progress reply message UA A generates PRACK (Provisional Response ACKnowledgement) message indicates the receipt of the 183 Session Progress response message. From this point, both of the sides start resource reservation as mentioned above. When UA A finishes reserving resources in the A->B direction, it sends an UPDATE to UA B. UA B returns a 200 (OK) response for the UPDATE, indicating that all the preconditions for the session have been met. indicating that all the preconditions for the session have been met.On the figure above, User Agent (UA) A sends INVITE message includes quality of service preconditions in the SDP. After receiving 183 Session Progress reply message UA A generates PRACK (Provisional Response ACKnowledgement) message indicates the receipt of the 183 Session Progress response message. From this point, both of the sides start resource reservation as mentioned above. When UA A finishes reserving resources in the A-> B direction, it sends an UPDATE to UA B. UA B returns a 200 (OK) response for the UPDATE, indicating that all the preconditions for the session have been met. indicating that all the preconditions for the session have been met.
IV. Implementation and ExperimentsIV. Implementation and Experiments
4.1 Implementation of SIP4.1 Implementation of SIP
The SIP agent has been implemented with Object Oriented design concepts. The SIP is referred in RFC 3261. Especially the SIP agent design and implement for interaction with RSVP-TE(fig. 7). In this paper, SIP proxy server use VOCAL system without any modification in it. The SIP agent has been implemented with Object Oriented design concepts. The SIP is referred to in RFC 3261. Especially the SIP agent design and implement for interaction with RSVP-TE (fig. 7). In this paper, SIP proxy server use VOCAL system without any modification in it.
(fig. 7) Class diagram for SIP user agent(fig. 7) Class diagram for SIP user agent
4.2 Implementation of RSVP-TE4.2 Implementation of RSVP-TE
In this section the Implementation of RSVP Server & RSVP client modules for UNI signaling are explained. The RFCs 2205 and 3209 give details of RSVP and RSVP-TE respectively. RSVP function modules has 2 parts : (i) management part, and (ii) message part. The (fig. 8) shows UML Class diagram of management part.In this section the Implementation of RSVP Server & RSVP client modules for UNI signaling are explained. The RFCs 2205 and 3209 give details of RSVP and RSVP-TE respectively. RSVP function modules has 2 parts: (i) management part, and (ii) message part. The (fig. 8) shows UML Class diagram of management part.
(fig. 8) Class diagram for RSVP-TE application(fig. 8) Class diagram for RSVP-TE application
And (fig. 9, 6) show implementation of RSVP's RESV and PATH message format respectively.And (fig. 9, 6) show implementation of RSVP's RESV and PATH message format respectively.
(fig. 9) RSVP Path Message formats(fig. 9) RSVP Path Message formats
(fig. 10) RSVP Resv Message formats(fig. 10) RSVP Resv Message formats
The RSVP protocol ID is 46 and it uses RAW socket to transmit data through RAW IP datagram. RSVP-TE Server handles RSVP resource reservation and resource allocation through the NMS. In each established session,The RSVP protocol ID is 46 and it uses RAW socket to transmit data through RAW IP datagram. RSVP-TE Server handles RSVP resource reservation and resource allocation through the NMS. In each established session,
the RSVP client sends one PATH message for resource reservation and one RESV message to allocate resource. To keep the reservation state persistent, a timer has been implemented which periodically sends PATH and RESV messages. For reservation tear down, the RESV TEAR and PATH TEAR messages have been implemented.the RSVP client sends one PATH message for resource reservation and one RESV message to allocate resource. To keep the reservation state persistent, a timer has been implemented which periodically sends PATH and RESV messages. For reservation tear down, the RESV TEAR and PATH TEAR messages have been implemented.
4.3 Multimedia coordinator4.3 Multimedia coordinator
Multimedia co-ordinator in User agent manages SIP and RSVP-TE modules and Audio-Video connections. Media co-ordinator manages details of Session ID, Source address, Source port number, Destination address, Destination port number, Protocol ID, Service class, Bandwidth etc..Multimedia co-ordinator in User agent manages SIP and RSVP-TE modules and Audio-Video connections. Media co-ordinator manages details of Session ID, Source address, Source port number, Destination address, Destination port number, Protocol ID, Service class, Bandwidth etc.
(fig. 11) shows a SIP User Agent's Register Invite Message GUI and the following figure shows User Agent of the receiver. Once invite message is accepted, the resource reservation will be carried out.(fig. 11) shows a SIP User Agent's Register Invite Message GUI and the following figure shows User Agent of the receiver. Once invite message is accepted, the resource reservation will be carried out.
(fig. 11) RSVP Resv Message formats(fig. 11) RSVP Resv Message formats
We generated network traffic through traffic generator and tested best effort and QoS-guaranteed services in our lab. From the above (fig. 12), it has been clear that the Qos-guaranteed service does not suffer form packet loss.We generated network traffic through traffic generator and tested best effort and QoS-guaranteed services in our lab. From the above (fig. 12), it has been clear that the Qos-guaranteed service does not suffer form packet loss.
(fig. 12) Comparison of QoS Guaranteed and Best Effort Services(fig. 12) Comparison of QoS Guaranteed and Best Effort Services
V. ConclusionV. Conclusion
In this paper we propose to use the existing SIP with RSVP-TE for implementing the guaranteed QoS between the user agents. In the proposed architecture, SIP and RSVP-TE is effectively used.In this paper we propose to use the existing SIP with RSVP-TE for implementing the guaranteed QoS between the user agents. In the proposed architecture, SIP and RSVP-TE is effectively used.
본 발명에 의하면 기존의 SIP(Session Initiation Protocol)/SDP(Session Description Protocol)를 이용하여 단말간의 Session를 설정하고, Session 설정 시 단말용 RSVP-TE agent를 이용, ISP의 RSVP-TE Server/CAC(Call Admission Control)과 UNI (User-Network Interface) Signaling을 통하여 QoS 보장형 Packet flow를 설정함으로써 End-To-End간 QoS를 보장해 준다. According to the present invention, a session between terminals is established using an existing Session Initiation Protocol (SIP) / Session Description Protocol (SDP), and an RSVP-TE server / CAC (ISP) of an ISP is used when a session is set up. QoS is guaranteed between end-to-end by setting QoS guaranteed packet flow through Call Admission Control) and UNI (User-Network Interface) signaling.
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