P2P architecture for IP telephony using SIP

Transcription

1 P2P architecture for IP telephony using SIP Fan Pu Helsinki University of Technology Abstract Session Initiation Protocol (SIP) is a control protocol standardized by IETF and implemented on the application layer. It can be used to establish, modify, and terminate Internet conferencing, telephony and some other multimedia sessions.[1] SIP uses server and client architecture, Peer-to- Peer (P2P) is another popular technology that accomplishes the same functionality in an overlay network where participants are organized as equally important peers. This paper gives a brief introduction to the motivation for P2P-SIP and the requirements for setting up its structure. Secondly, two design alternatives of P2P-SIP architectures are introduced based on the current level. It explains how the SIP-based IP telephony system utilizes the P2P architecture. The paper concludes with a brief look at the limitations of the P2P Internet telephony using SIP and suggests further work is required. KEYWORDS: SIP, P2P, P2P-SIP overlay, DHT 1 Introduction SIP-based Internet Telephony is widely used by various users and organizations. Unfortunately, along with the increased adoption, more problems arise. A primary cause of them is that SIP uses centralized servers in its network. This structure negatively affects the scalability of SIP. So it makes sense to introduce P2P technologies to attack these problems. P2P, as a self-organized architecture, does not require any server in the network and hence makes a good solution to those problems mentioned in the above. This paper discusses how P2P can be implemented on SIP-based Internet Telephony systems. Firstly, it gives background information about Internet Telephony, SIP, P2P and the related studies. The paper examines the rules currently implemented for SIP, then analyses the necessity of introducing P2P into SIP-based Internet Telephony systems in Sec. 2. Next, in Sec. 3, it gives introduction to the P2P technology and its popular applications. In addition, it illustrates the architectures and algorithms used in P2P networks and the advantages they bring to SIP-based Internet Telephony. Sec. 4 describes two design alternatives proposed by P2P SIP working group, and compares the feasibility of these two solutions by studying current implementations. In addition, in Sec. 5 this paper describes a P2P-SIP overlay scheme in details. This description includes which algorithms are used, how the node- and user-level operations proceed. Sec. 6 gives conclusion and present security consideration as future work for P2P-SIP overlay. 2 Overview of SIP 2.1 Terminologies involved in Internet Telephony and SIP Presence User Agent (PUA): An application is used to generate information of user s presence state and push this presence information to presence agent (PA).[4] Presence Agent(PA): A kind of user agent in the SIP network, it can accept SUBSCRIBE requests from subscribers and respond to them. It also can generate NOTIFY message to inform subscribers of changes in presence status.[4] Registrar: Registrar is a server, it is used to accept REGIS- TER requests from clients and forward the information they contain to a location service.[2] Proxy Server: It is an intermediate entity in the SIP network. It can act as a server to accept SIP requests or act as a client to forward SIP requests.[2] User Agent Server: It represents users receiving SIP requests on their behalf and giving response to these SIP requests.[2] 2.2 Internet Telephony and Session Initiation Protocol IP telephony or voice over IP means using IP as a foundation for telephone service. Traditional Switched Public Telephone Network (SPTN) requires a signaling mechanism, known as Signalling System 7 (SS7)[13], to establish connection before audio is sent. The same as PSTN, IP telephony system also needs this kind of mechanism. It is known as signaling protocols which are used to establish media session in IP telephony. Only after initiating multimedia session successfully, could other protocols, such as actually transporting media, be used for IP telephony. At present, two standards for IP telephony exist. One is known as H.323[12], a suite of protocols, which specifies how to form a functional IP telephony system by using multiple protocols. The other standard is SIP, which only covers the signaling of IP telephony. However, in order to accomplish the integrate IP telephony system, besides SIP, other protocols are also required, such as Session Description Protocol (SDP)[14], Real-time Transport Protocol (RTP)[15], RTP Control Protocol (RTCP)[15], to support media and transport parameters for session, real-time multimedia transport, etc. In this paper, we only concentrate on the aspect of

2 multimedia session initiating for IP telephony system, independent of transport. SIP uses a client-server model. In SIP network, on the server side, it contains proxy server, redirect server, user agent server and registrar. On the client side, it contains user agent client and proxy client. SIP uses SIP URI to identify the user s public address, which represents users at hosts. The use part of SIP URI could be a user name or a telephone number. The host part of SIP URI could be a domain name or a numeric network. For example, a SIP address is sip: alice@aaa.com. When a caller wants to initiate a session to a callee, he first needs to know the SIP address of callee. The process is as following: caller could either send a "INVITE" request through caller s locally configured SIP proxy server, or directly send to the IP address and port corresponding to the callee s address. Via the callee s location service, the caller can find a more precise location of callee. After callee s responsible proxy server forwards this request to User Agent (UA) and User Agent Server (UAS) of callee alerts, it returns a successful message to the caller. Then the caller confirms this message and sends an "ACK" request to callee, the session is initialized successfully. An important function SIP provides is user mobility with "REGISTER" transaction. It means if a user wishes to receive a call, he should register himself with user agent to registrar server first. In this case, his SIP proxy server could forward the message to user agent according to the registered address. SIP also allows a user to register several user agents to his registrar server. For example, a user can register his home phone, office phone and mobile phone to registrar at the same time. Thus, by using location service, proxy could reach these three client agents simultaneously. SIP can perform presence service of users; let them know the target user s presence states in the network. This function is implemented by "NOTIFY" and "SUBSCRIBE" methods. User utilizes PUA to push data about presence information into the PA. Usually PA is co-locating with proxy or registrar server. For instance, when user A wants to know presence information about user B, user A is first required to generate a "SUBSCRIBE" request which contains B s SIP URI. Just like SIP could route A s "INVITE" request to B s proxy server, SIP is also capable to route A s "SUBSCRIBE" request to B s proxy server. Since B s proxy server has already known B s presence status, it is able to send "NOTIFY" message to user A after A is authorized successfully. The "NO- TIFY" message can have more detailed specifications, such as "on line, but needs coffee" etc. In addition, whenever user B changes its state, B s PA always sends new "NOTIFY" message to all the subscribers who have already been authorized by B. 2.3 Limitations of SIP Scalability: SIP uses server-client interaction. While a lot of users join SIP network, the servers get heavy strain to respond to requests from the clients. Because availability is unscalable for centralized network, we need decentralized system to be more reliable and scalable. Security Restriction: "Services typically used by individuals and small organizations, such as AOL s AIM. Microsoft s Messenger, and the Vonage VoIP service, user servers located at the provider."[11]. As a small organization, if it does not allow internal communications to flow outside to the external party for security restriction reason, or, it costs too much to build its own internal system, it will have problem. Complex Maintenance and Configuration: Maintenance and configuration is needed both at server and client for SIP network. For example, UA may need to be configured to find registrar s address during registration process. Registrar needs to be configured to bind the sets of domain which needs its maintenance. Therefore, redundant maintenance and configuration lead to Ad-hoc and ephemeral groups are not so easy to be established by using SIP network. Controlled Infrastructure: If Internet infrastructure is damaged in civil emergency, or Internet Service Provider (ISP) can not provide Internet access successfully, ISP clients could not connect to server. Thus SIP service is impossible to obtain even between connected nodes. This is another limitation of SIP which motivates P2P architecture to give solution. Restriction of Usability: A few users may have restrictions for connecting to persistent, centralized servers due to government censorship or because of ISP limitations to competing service. In this case, a network that does not require servers is needed. 3 Overview of P2P 3.1 Terminologies involved in P2P SIPPING Working Group defines following terms for P2P Overlay Network and DHT: Overlay (Network) or P2P Overlay (Network): A logical network is formed by the nodes which implement the same P2P service or application.[8] "Distributed Hash Table (DHT): A mechanism in which resources and given a unique key produced by hashing some attribute of the resource, locating them in a hash space. Nodes located in this hash space also have a unique id within the hash space. Nodes store information about resources with keys that are numerically similar to the node s ID in the hash space."[9] 3.2 Basic Concept of P2P and its applications P2P system is another model different from client-server model. Every communication entity in the network plays an equally important role, which is named peer or node. Node has both client and server features. Therefore, a node in network is not only a client, but also provides functionality as servers to respond to other clients.[6] P2P is usually classified into two kinds of system.[7] One is known as unstructured system. In this system, the nodes

3 super nodes in DHT ordinary nodes Figure 1: DHT structure with super nodes and ordinary nodes have the same capability to serve other clients. And the network does not have any logical structure. There is one distinct character of this system: while node wants to search other resource or obtain any service, it will flood the request to all of its neighbour nodes.[6] Therefore, it is an inefficient P2P system and will not be utilized in the P2P SIP architecture. In [11], it mentions that one application named WASTE uses unstructured P2P system. WASTE is an application used for file sharing and Instant Message. The node running WASTE uses the same encryption key to encrypt traffic in the same overlay and sends request by flooding message to all other nodes. By Contrast, the other kind of system is structured system. In this system, some nodes may have more powerful CPU, memory, network bandwidth and has public IP address outside network address translation (NAT), so they have more ability to provide functionality to serve other nodes. These nodes are referred as super nodes.[7] In structured system, only super nodes form DHT in order to locate resource more efficiently.[3] Unlike super nodes, some nodes in the network have less capacity and availability. For instance, the nodes behind firewall or NATs, could become the other type named an ordinary node.[7] The ordinary node looks up other nodes in the network via connecting super node to find other node s location. Fig. 1 shows the structure of this system. The most popular VoIP application Skype uses this system. In [5], it describes by using host cache, Skype client stores IP addresses of super nodes in PC and use chord which is a kind of algorithms maintaining DHT to quickly find the other nodes. 3.3 P2P Architectures One important point of P2P network is that nodes are ephemeral. It means nodes can join or leave the P2P network at any time and it requires P2P architecture has ability to adapt to the dynamic changes all the time. Currently, many P2P architectures that we know are designed according this principle. In this subsection, it does not specify all the existing P2P architectures, but only the DHT system and chord algorithm. They are used in P2P-SIP overlay that will be studied in Sec. 5. DHT is used to locate resources.[6] As mentioned in Sec. 3.2, only super nodes build DHT. In the DHT, every resource has a resource-id, which is calculated by hashing some keyword or value of the resource in question. This resource-id is unique in the DHT and identifies its corresponding resource. Furthermore, each node in the DHT is also assigned an identifier, namely Node-ID, by performing some hash operation. The resulting node-id is also unique in the DHT. It should be noted here that both the resource-ids and the node-ids share the same hash space. In hash space, a node with node-id always has some resource-ids which are close to it. By using the Node-ID, this particular node stores the resources as entities in a hash table with resource-id. In this manner, this particular node is the key of the resources stored in it. It means that, in order to find the target resource, a requester only needs to contact the node storing target resource with Node-ID. If the node is responsible for the target Resource-ID and know how to find the resource, it will generate a reply to the requester. In addition, the node with node-id may leave or enter the overlay at any time. Thus, it requires the node be capable to exchange the message and stored entities to prevent from losing information in DHT. One famous algorithm based on DHT is called Chord.[6] It uses ring-type structure to determine the node s place. In chord, each node has a finger table to store certain amounts of entities. According to the entity s number in its finger table, it maps particular entity as neighbor node in the same hash space. To find the target node using chord, it can be achieved by starting search the node in the finger table which is the closest to the target node. Since each node knows more about nearby nodes than nodes farther way, the intermediate neighbor nodes always get more precise location information about the target node. After repeating searches several times, the target node will be found eventually. 3.4 Advantages of P2P Primarily two advantages may be introduced from P2P for SIP-based telephony: Scalability: When every node joins the network, it has potential to provide functionality as a server. Therefore the amount of servers is increased when the number of nodes has increased in the network and the tasks performed by the server will not concentrate on some fixed nodes. In this fashion, unlike server-client model, it does not affect much about scalability along with the increasing network. Robustness: In structured P2P network, it has several super nodes that are active as servers in SIP network. Ordinary node could get response from every super node it is connecting to. Therefore the communication between ordinary node and the other nodes can not fail in case of an unsuccessful super node. From this

4 view, P2P network increases the robustness comparing with pure SIP network. FIND P2P network INSERT 4 Design Alternatives for P2P and SIP Currently several approaches have been proposed about P2P architectures for SIP-based IP telephony. These proposals can be classified into two categories: one is SIP using P2P, which uses P2P protocols to implement SIP location service. The other is P2P over SIP. It uses SIP messages to transport P2P traffic. In this section, we discuss these two design alternatives and their common requirements. 4.1 Common Requirements of Both Alternatives Since different requirements could result in different design schemes, we need to identify the design goals for the combined P2P and SIP network first. In [9], the author defines a set of requirements for P2P-SIP. This subsection picks some general points from his paper as design goals for P2P SIP network. First, new P2P and SIP network should fulfill the functions and services traditional SIP network could provide. For example, it can support the establishment, modification and termination of basic voice call, multimedia conference, etc. Secondly, it could introduce some P2P s characters into the network which is absent in SIP network. Concretely, the centralized resource, such as registrar, proxy server in SIP network should be decentralized to distributed peers. Thirdly, new network should not break the existing service and protocols which are in original P2P network or SIP network. For instance, protocols like SSL, TLS should still be used; NAT and firewall traversal also need to be supported in the new network system the same way as in the P2P system. 4.2 SIP using P2P One design alternative is SIP using P2P, as shown in Fig. 2. It is proposed by A.Johnston in [10]. In this design, it makes P2P protocol replace SIP location service. Instead of using SIP message, such as REGISTER, P2P client, who acts as SIP user agent, directly inserts its URI or real time URL into DHT and it is stored as particular search key. Furthermore, in order to accomplish the location service as SIP proxy or direct server should get precise information to forward the message to the target user agent; P2P protocols can provide the necessary suite of functions. They include looking up the search key in DHT, updating and publishing the data information related to this search key, etc. Associated with these functions, the location service can be carried out with P2P protocols in this design. In [10], the author believes that this design is a good starting point for distributed location service, which makes it efficient to look up the resource location. However, it still has some issues that need to be considered in the real implementation. One important issue is that, the nodes formed in this network can not be working for NAT Traversal at the same time. Moreover, the P2P protocols used for this design can not handle the NAT Traversal very well. The author has not INVITE sip: alice@aaa.com Figure 2: Alternative of SIP using P2P INVITE alice P2P-SIP overlay REGISTER Figure 3: Alternative of P2P over SIP Alice sip: alice@aaa.com Alice sip: alice@aaa.com explained why NAT traversal issue is a problem in this design. However, if a node behind NAT wants to join the network successfully, it should obtain a NAT Traversal relay first. 4.3 P2P over SIP The other design alternative is P2P over SIP, as shown in Fig. 3, which implies implementation of P2P using SIP message. In detail, SIP message like REGISTER, INVITE are still used. It may alter the header of original SIP message in order to accomplish more functions, like indicating nodes joining or leaving the overlay. For example, one kind of P2P- SIP architecture[7] proposed by Kundan Singh and Henning Schulzrinne employs this kind of design. It uses "Require" header for OPTIONS or REGISTER to extend the original SIP message. This design is easy to be developed since SIP is mature, using SIP as foundation and carry out P2P traffic on top of it will not cost extra stack or require much change of SIP itself. In the following section, we will put more attention to introduce this design, especially the P2P-SIP architecture mentioned above, instead of SIP using P2P. 5 The Architecture of P2P-SIP overlay We present a P2P-SIP architecture proposed by Kundan Singh and Henning Schulzrinne[7] as a case study of P2P over SIP approach. In the first subsection, we look into its structure and messages, then in the next, its node operations, and finally in the last subsection, we study its user-level operations. 5.1 Structure and Message This P2P-SIP architecture defines SIP as "underlying protocol for location user, registering the user, call setup and instant message."[7] Besides, it has DHT like Chord to provide

5 user location information to each node in its overlay. The same as structured P2P network, the nodes in this architecture consists of ordinary nodes and super nodes. The nodes which have public address and high capability is qualified to become super node. Other nodes are belongs to ordinary nodes. The nodes in DHT may use "find", "join", and "leave" methods. Nodes use SIP message to communicate in this architecture. The SIP message used includes: REGISTER, OP- TIONS, INVITE, REFER, etc. Besides making multimedia call, it also uses SIP MESSAGE to transfer instant message. 5.2 Node Operations In this subsection, we will describe three procedures: 1. a node enters the network. 2. a node leaves the network. 3. a node encounters failure in the network. When a node enters the network, via SIP URI, such as sip: alice@aaa.com, node will register itself using traditional SIP REGISTER method and P2P mechanism simultaneity. By using traditional SIP REGISTER method, the node uses DNS to find possible SIP server to register it with REG- ISTER message. Besides, by using P2P mechanism, node could use multicast way with limited time-to-live (TTL) values to find the nearest super node in order to join the P2P overlay. Once the super node or any peer addresses are known before and cached in the local machine of the node, it will prefer to contact these addresses whenever it starts up again. Besides these methods, in [7], it also introduces some other methods, like pre-configure bootstrap peers in the node to be the first contacting peers when node starts to join the P2P overlay. When a node successfully leaves the network, we should consider whether the node is an ordinary node or a super node. In the first case, the ordinary node simply needs to send an un-register message to its connected super node to inform that it will leave the network. After that, the responsible super node will propagate this information to relevant super nodes in DHT, which have stored the left ordinary node as a key. In the latter case, the leaving super node needs to use REGISTER message to transfer the data of all its responsible ordinary nodes to other super nodes in the DHT. This way, the information of the ordinary nodes attached to this leaving super node will not be lost. These ordinary nodes will later find that they have failed to connect to the previous super node and that they have been transferred to the new super nodes when they refresh registration next time. When a node encounters failure suddenly in the network, the following operations also depend on whether the node is ordinary or super node. If the node is an ordinary node, only the super node it connects to will know its failure since the broken node will stop refreshing its registration. After that, the super node will send an OPTION message to this node and observe any response to confirm this failure. If the broken node is a super node, its neighbour nodes in DHT will detect its failure and try to be responsible for its ordinary nodes. However, unless the ordinary nodes refresh REGIS- TER message to new super node or the new super node already has known their information, the ordinary nodes connection may be lost after the failure of their original super node. 5.3 User-level Operations In this subsection, we will give short description on how the user sets up a call and obtain presence information. First we consider the process of a user setting up a call. In traditional SIP network, caller will start to send INVITE message to contact callee to initiate a session. In this P2P- SIP overlay, the difference for this process is that, caller needs to find the node location of the callee with INVITE message. At the beginning, caller will check the callee s address if it is used recently. If the address is cached in local machine, the caller will use it first. Nevertheless, the client using this address may not respond because of absence, and then this address will be deleted from cache list. In case that the caller cannot find cached address of callee, he will start the SIP-based look up by querying of DNS and P2Pbased look up simultaneously. When it uses P2P-based look up, the overlay uses DHT to locate the destination node of callee. After mapping to destination node successfully, the super node responsible for caller will acts like SIP proxy or redirect server to forward message to callee. The decision of this super node acting as proxy or redirect server depends on whether the node of caller behind firewall or NAT. If the caller as ordinary node is behind firewall or NAT, the super node will act as SIP proxy. Otherwise, it prefers to work as redirect server to redirect INVITE message to the next hop. Afterward, the session is initialized and user can start the end-to-end multimedia transmission like traditional SIP network being. Like SIP can perform presence service for users, in this P2P-SIP overlay, user can obtain other user s presence status too. User can have a buddy list together with his identifier. As profile information of the user, this buddy list can be encrypted and stored in the intermediate nodes in P2P- SIP network. Whenever the user sign in with his identifier, act as a node joins into the overlay, he can fetch his profile information and start to locate the users in buddy list. In [7], it does not specify if the SUBSCRIBE and NOTIFY SIP messages are used for presence service in P2P-SIP overlay, If this mechanism is used, it should be combined with P2Pbased look up or SIP-based look up to locate the destination node. 6 Conclusion and Future Work First of all, this paper studies the background knowledge of SIP based IP telephony and pure P2P architecture. We gain some idea behind them, which is P2P s inherent character like good scalability, free configuration which are useful for SIP network. Thus we present some design alternatives that place P2P and SIP together. Moreover, as an easily deployed solution, this paper specifies one P2P-SIP overlay proposed by P2P SIP working group. It explains how to implement the

6 overlay from structures, message, node operations and user level operations. As far as future work is concerned, the network combining P2P and SIP also give rise to a series of open issues. Security is one important issue for future work. Since the nodes in P2P network is not as trusted as the servers in SIP network, some security mechanism needs to be uses to protect call information during routing call message in the network. Some security solutions are given for P2P system. However, those focus on no-real-time system such as file sharing applications. As a result, more work is needed to solve this problem for P2P SIP network. In addition, there are still some other open issues need to be done, such as "free riding" threat specified in [7], how to enable the nodes behind firewall and NATs to become as super node in order to reduce the work load of the nodes with public address, etc. Summing up, P2P architecture for SIP brings big advantages and will act as powerful role in IP telephony in future. References [1] M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg. SIP: Session Initiation Protocol RFC 2543, Internet Engineering Task Force, March URL: [2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler. SIP: session initiation protocol RFC 3261, Internet Engineering Task Force, June URL: [3] E. Shim, S. Narayanan, G. Daley. An Architecture for Peer-to-Peer Session Initiation Protocol (P2P SIP) IETF Internet Draft, February, URL: [8] David A. Bryan, E. Shim and Bruce B. Lowekamp. Use Cases for Peer-to-Peer Session Initiation Protocol (P2P SIP) IETF Internet Draft, November 29, URL: [9] S. Baset, H. Schulzrinne, E. Shim and K. Dhara. Requirements for SIP-based Peer-to-Peer Internet Telephony IETF Internet Draft, October, URL: salman/drafts/draftbaset-sipping-p2preq-00.txt [10] A.Johnston. SIP, P2P, and Internet Communications IETF Internet Draft, March, URL: [11] David A. Bryan, Bruce B. Lowekamp, Cullen Jennings. SOSIMPLE: A Serverless, Standards-based, P2P SIP Communication System Appears in AAA-IDA 2005 IEEE URL: bryan/pubs/bryan- AAA-IDEA2005.pdf [12] Packetizer, Inc. H.323 Informatoin Site, URL: [13] Performance Technologies, Inc. SS7 Tutorial, URL: [14] M. Handley, V. Jacobson. SDP: Session Description Protocol RFC 2327, Internet Engineering Task Force, April URL: [15] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson. RTP: A Transport Protocol for Real-Time Applications RFC 1889, Internet Engineering Task Force, January URL: [4] J. Rosenberg. A Presence Event Package for the Session Initiation Protocol (SIP) RFC 3856, Internet Engineering Task Force, August URL: [5] Salman A. Baset, H. Schulzrinne. An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol Columbia University, New York, NY, Tech. Rep. CUCS , September 15, URL: library/trrepository/reports/reports-2004/cucs pdf [6] David A. Bryan, Bruce B. Lowekamp, and C. Jennings. A P2P Approach to SIP Registration IETF Internet Draft, March, URL: 02.txt [7] K. Singh and H. Schulzrinne. Peer-to-Peer Ineternet Telephony using SIP Columbia University Technical Report CUCS , New York, NY, Oct URL: library/trrepository/reports/reports-2004/cucs pdf