SIP WG R. Mahy Internet-Draft SIP Edge LLC Expires: August 5, 2006 V. Gurbani, Ed. Lucent Technologies, Inc./Bell Laboratories B. Tate BroadSoft February 2006 Connection Reuse in the Session Initiation Protocol (SIP) draft-ietf-sip-connect-reuse-05.txt Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on August 5, 2006. Copyright Notice Copyright (C) The Internet Society (2006). Abstract When SIP entities use a connection oriented protocol to send a request, they typically originate their connections from an ephemeral port. The SIP protocol includes mechanisms which insure that responses to a request, and new requests sent in the original Mahy, et al. Expires August 5, 2006 [Page 1] Internet-Draft SIP Connection Reuse February 2006 direction reuse an existing connection. However, new requests sent in the opposite direction are unlikely to reuse the existing connection. This frequently causes a pair of SIP entities to use one connection for requests sent in each direction, and can result in potential scaling and performance problems. This document proposes requirements and a mechanism which address this deficiency in environments where the connection could be opened in either direction. Table of Contents 1. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Applicability Statement . . . . . . . . . . . . . . . . . . 3 3. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 4. Benefits of Connection Reuse . . . . . . . . . . . . . . . . 5 5. Overview of Operation . . . . . . . . . . . . . . . . . . . 6 6. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 8 7. Formal Syntax . . . . . . . . . . . . . . . . . . . . . . . 8 8. Normative Behavior . . . . . . . . . . . . . . . . . . . . . 8 8.1 Client Behavior . . . . . . . . . . . . . . . . . . . . . 9 8.2 Server Behavior . . . . . . . . . . . . . . . . . . . . . 10 9. Security Considerations . . . . . . . . . . . . . . . . . . 11 9.1 Authenticating TLS Client Connections . . . . . . . . . . 11 9.2 Authenticating TLS Server Connections . . . . . . . . . . 11 9.3 Security Considerations for the TCP Transport . . . . . . 11 10. Connection Reuse and SRV Interaction . . . . . . . . . . . . 13 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . 13 12. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . 14 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 14 13.1 Normative References . . . . . . . . . . . . . . . . . . 14 13.2 Informational References . . . . . . . . . . . . . . . . 14 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 15 Intellectual Property and Copyright Statements . . . . . . . 16 Mahy, et al. Expires August 5, 2006 [Page 2] Internet-Draft SIP Connection Reuse February 2006 1. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [3]. Additional terminology used in this document: Advertised address: The address that occurs in the Via sent-by production rule, including the port number and transport. Alias: A transport layer connection associated with a resolved address. Resolved address: The address of a user agent (including port number and transport) retrieved from the DNS resolution contained in RFC3263 [5]. 2. Applicability Statement The applicability of the mechanism described in this document is for two adjacent SIP entities to reuse connections when they are agnostic about the direction of the connection, i.e., either end can initiate the connection. SIP entities that can only open a connection in a specific direction -- perhaps because of Network Address Translation (NAT) and firewall reasons -- reuse their connections using the mechanism described in [1]. The connect reuse mechanism described in this document is defined only for Transport Layer Security (TLS) transports. Specifically, implementations MUST NOT use this mechanism for the TCP transport due to the possible attacks that can be launched with connection reuse over TCP. Such attacks and alternative methods for connection reuse over TCP are described in Section 9.3. 3. Introduction SIP [2] entities can communicate using either unreliable/ connectionless (e.g., UDP) or reliable/connection-oriented (e.g., TCP, SCTP [11]) transport protocols. When SIP entities use a connection-oriented protocol (such as TCP or SCTP) to send a request, they typically originate their connections from an ephemeral port. In the following example, Entity A listens for SIP requests over TLS [4] on TCP port 5061 (the default port for SIP over TLS over TCP), but uses an ephemeral port (port 8293) for a new connection to Entity B. These entities could be SIP User Agents or SIP Proxy Servers. Mahy, et al. Expires August 5, 2006 [Page 3] Internet-Draft SIP Connection Reuse February 2006 +-----------+ 8293 (UAC) 5061 (UAS) +-----------+ | |--------------------------->| | | Entity | | Entity | | A | | B | | | 5061 (UAS) | | +-----------+ +-----------+ Figure 1: Uni-directional connection for requests from A to B. The SIP protocol includes mechanisms which insure that responses to a request reuse the existing connection which is typically still available, and also includes provisions for reusing existing connections for other requests sent by the originator of the connection. However, new requests sent in the opposite direction -- in the example above, requests from B destined to A -- are unlikely to reuse the existing connection. This frequently causes a pair of SIP entities to use one connection for requests sent in each direction, as shown below. +-----------+ 8293 5061 +-----------+ | |.......................>| | | Entity | | Entity | | A | 5061 9741 | B | | |<-----------------------| | +-----------+ +-----------+ Figure 2: Two connections for requests between A and B. Opening an extra connection where an existing one is sufficient can result in potential scaling and performance problems. Consider the call flow shown below where Proxy A and Proxy B use the Record-Route mechanism to stay involved in a dialog. Proxy B will establish a new TLS connection just to send a BYE request. Mahy, et al. Expires August 5, 2006 [Page 4] Internet-Draft SIP Connection Reuse February 2006 Proxy A Proxy B | | Create connection 1 +---INV--->| | | |<---200---+ Response over connection 1 | | Re-use connection 1 +---ACK--->| | | = = | | |<---BYE---+ Create connection 2 | | Response over +---200--->| connection 2 Figure 3: Multiple connections for requests. Thus, it is advantageous to reuse connections whenever possible. 4. Benefits of Connection Reuse Opening an extra connection where an existing one is sufficient can result in potential scaling and performance problems. For example, each new connection using TLS requires a TCP 3-way handshake, a handful of round-trips to establish TLS, typically expensive asymmetric authentication and key generation algorithms, and certificate verification. This effectively doubles the load on each entity. Setting up a second connection (from B to A above) for subsequent requests, even requests in the context of an existing dialog (e.g., re-INVITE or BYE after an initial INVITE, or a NOTIFY after a SUBSCRIBE [10] or a REFER [9]), can also cause excessive delay (especially in networks with long round-trip times). ReINVITEs or UPDATE [7] requests are expected to be handled automatically and rapidly in order to avoid media and session state from being out of step. If a reINVITE requires a new TLS connection, the reINVITE could be delayed by several extra round-trip times. Depending on the round-trip time, this combined delay could be perceptible or even annoying to a human user. This is especially problematic for some common SIP call flows (for example, the recommended example flow in figure number 4 in RFC3725 [8] use many reINVITEs). The mechanism described in this document can mitigate the delays associated with subsequent requests. Mahy, et al. Expires August 5, 2006 [Page 5] Internet-Draft SIP Connection Reuse February 2006 5. Overview of Operation This section is tutorial in nature, and does not specify any normative behavior. The act of reusing a connection is initiated by an user agent client (UAC, or the proxy half of the UAC) when it adds an "alias" parameter to the added Via header (the parameter itself is defined later). When a user agent server (UAS, or the proxy half of the UAS) receives the request, it examines the topmost Via header. If the header contained an "alias" parameter, the UAS establishes a binding such that subsequent requests going to the UAC will reuse the connection. We now explain this working in more detail in the context of communication between two adjacent proxies. Without any loss of generality, it should be clear that the same technique can be used for connection reuse between a UAC and an edge proxy, or between an edge proxy and a UAS, or between an UAC and an UAS. P1 and P2 are proxies responsible for routing SIP requests through user agents that use them as edge proxies (see Figure 4). P1 P2 p1.example.com p2.example.com (192.0.2.1) (192.0.2.128) Figure 4: Proxy setup. This document is concerned with specifying an extension to SIP for connection reuse at the receiving end; i.e., reusing the connection when P2 wants to send a request to P1. However, it should be clear that P1 can reuse a connection previously established with P2. In fact, the SIP community recommends that clients reuse a connection previously established with a server for subsequent transactions going to the same resolved address. Thus, the reuse property of a connection, once it is established, is bi-directional and alias tables may be maintained at both P1 and P2. P1 gets a request from one of its upstream user agents, and after performing RFC3263 server selection, arrives at a destination address of P2. P1 maintains an alias table, and it populates the alias table with the IP address, port number, and transport of P2 as determined through RFC3263 server selection. P1 adds an "alias" parameter to the topmost Via header (inserted by it) before sending the request to P2. The value in the sent-by production rule of the Via header (including the port number), and the transport over which the request was sent becomes the advertised address of P1: Via: SIP/2.0/TLS p1.example.com;branch=z9hG4bKa7c8dze;alias Mahy, et al. Expires August 5, 2006 [Page 6] Internet-Draft SIP Connection Reuse February 2006 Assuming that P1 does not have an existing aliased connection with P2, P1 now opens a connection with P2. Upon connection authentication and acceptance, it adds P2s to its alias table. P1's alias table now looks like: Destination Destination Destination Alias Connection IP Address Port Transport Descriptor ... 192.0.2.128 5061 TLS 25 Subsequent requests that traverse from P1 to P2 will reuse this connection; i.e., the requests will be sent over the descriptor 25. When P2 receives the request, it may add a "received" parameter to the topmost Via and examines the topmost Via to determine whether P1 supports aliased connections. The Via at P2 now looks like: Via: SIP/2.0/TLS p1.example.com;branch=z9hG4bKa7c8dze;alias; received=192.0.2.1 The presence of the "alias" parameter indicates that P1 does support aliasing. P2 now authenticates the connection and if the authentication was successful, P2 creates an alias to P1 using the advertised address in the topmost Via. P2's alias table looks like: Destination Destination Destination Alias Connection IP Address Port Transport Descriptor ... 192.0.2.1 5061 TLS 18 There are two items of interest here: 1. Note that the entry in the last column for P2's alias table is the descriptor over which the connection was passively accepted. When P2 gets a request from one of its user agents, and determines through RFC3263 server resolution that the request should be sent to P1 over TLS using the default port (5061), it will reuse the aliased connection accessible to it through descriptor 18 instead of opening a new connection. 2. The network address inserted in the "Destination IP Address" column should be the source address as seen by P2 (i.e., the "received" parameter). It could be the case that the host name of P1 resolves to different IP addresses due to round-robin DNS. However, the aliased connection is to be established with the original sender of the request. To implement connection aliases for resolved addresses, a SIP node could (for example) search an additional data structure (the alias table) prior to opening a new connection, or could modify the data Mahy, et al. Expires August 5, 2006 [Page 7] Internet-Draft SIP Connection Reuse February 2006 structure in which it keeps active connection state so that aliases, active connections, and blacklisted nodes are all discovered when looking for an active connection. 6. Requirements 1. A connection sharing mechanism SHOULD allow SIP entities to reuse existing connections for requests and responses originated from either peer in the connection. 2. A connection sharing mechanism MUST NOT require UACs (clients) to send all traffic from well-know SIP ports. 3. A connection sharing mechanism MUST NOT require configuring ephemeral port numbers in DNS. 4. A connection sharing mechanism MUST prevent unauthorized hijacking of other connections. 5. Connection sharing SHOULD persist across SIP transactions and dialogs. 6. There is no requirement to share a complete path for ordinary connection reuse. Hop-by-hop connection sharing is more appropriate. 7. Formal Syntax The following syntax specification uses the augmented Backus-Naur Form (BNF) as described in RFC 4234 [6]. This document extends the via-params to include a new via-alias defined below. via-params = via-ttl / via-maddr / via-received / via-branch / via-alias / via-extension via-alias = "alias" 8. Normative Behavior This document specifies how to reuse connections. The SIP community recommends that servers keep connections up unless they need to reclaim resources, and that clients keep connections up as long as they are needed. Connection reuse works best when the client and the server maintain their connections for long periods of time. SIP entities therefore SHOULD NOT automatically drop connections on completion of a transaction or termination of a dialog. An alias is formed at the receiver of a request when it gets a request with the "alias" parameter in the topmost Via header. If the receiver decides to accept the alias, then the alias corresponds to the source IP address, transport, and port (if one exists in the Via sent-by, or the default port if it does not) of the sender of the Mahy, et al. Expires August 5, 2006 [Page 8] Internet-Draft SIP Connection Reuse February 2006 request. Whenever the RFC3263 server selection mechanism executed at the receiver results in the choice of this IP address, port, and transport tuple, the alias MUST be used instead. Note that at the receiver, the responses are sent over the same connection as specified by RFC3261. The aliasing mechanism at the receiver allows subsequent requests going from the receiver to the original sender of the request to reuse the same connection. An alias is formed at the sender of the request when it executes the RFC3263 server selection mechanism to arrive at an IP address, port, and transport tuple to send a request to. Subsequent requests going to the same destination address MUST use the alias instead. Only one alias SHOULD exist for the resolved address. If more than one alias is requested because of race conditions (or any other reasons), the receiver SHOULD consider the latest alias to be the desired alias. The receiver MUST NOT interpret the situation as a desire for load balancing between the aliases. Because an alias connection might be reclaimed during a transaction, clients SHOULD NOT enforce the RFC 3261 requirement of sending CANCEL and ACK (for non 2xx responses) to the same port. If the alias connection no longer exists, the client SHOULD open a new connection to the resolved address and send the CANCEL or ACK there instead. The newly opened connection MAY be inserted into the alias table. 8.1 Client Behavior The proposed mechanism uses a new Via header field parameter. The "alias" parameter is included in a Via header field value to indicate that the client wants to create a transport layer alias. The client places its advertised address in the Via header field value (in the "sent-by" production). The implications of placing an "alias" parameter in the topmost Via header of a request must be understood by the client. Specifically, this means that the client MUST keep the connection open for as long as the resources on the host operating system allow it to, and that it MUST accept requests over this connection -- as opposed to a default listening port -- from its downstream peer. And furthermore, it MUST reuse the connection when subsequent requests in the same or different transactions are destined to the same resolved address. Note that RFC3261 states that a response should arrive over the same connection that was opened for a request. Whether or not to allow an aliased connection ultimately depends on Mahy, et al. Expires August 5, 2006 [Page 9] Internet-Draft SIP Connection Reuse February 2006 the recepient of the request. Thus, clients MUST NOT assume that the acceptance of a request by a server automatically enables connection aliasing. They MUST continue receiving requests on their default port. Clients must be prepared for the case that the connection no longer exists when they are ready to send a subsequent request over it. This may happen if the peer ran out of operating system resources and had to close the connection. In such a case, a new connection MUST be opened to the resolved address and the alias table updated accordingly. Clients must authenticate the connection before forming an alias. Section 9.1 discusses the authentication steps in more detail. 8.2 Server Behavior When a server receives a request whose topmost Via header contains an "alias" parameter, it signifies that the upstream client will leave the connection open beyond the transaction and dialog lifetime, and that subsequent transactions and dialogs that are destined to a resolved address that matches the identifiers in the advertised address in the topmost Via header can reuse this connection. Whether or not to honor an aliased connection ultimately depends on the policies of the server. It MAY choose to honor it, and thereby send subsequent requests over the aliased connection. If the server chooses not to honor an aliased connection, it MUST allow the request to proceed as though the "alias" parameter was not present in the topmost Via header. This assures interoperability with RFC3261 server behavior. Clients should feel comfortable including the "alias" parameter without fear that the server will reject the SIP request because of its presence. Servers MUST be prepared to deal with the case that the aliased connection no longer exist when they are ready to send a subsequent request over it. This may happen if the peer ran out of operating system resources and had to close the connection. In such a case, a new connection MUST be opened to the resolved address and the alias table updated accordingly. If the Via sent-by contains a port, it MUST be used as a destination port. Otherwise the default port is the destination port. Servers must authenticate the connection before forming an alias. Section 9.2 discusses the authentication steps in more detail. Mahy, et al. Expires August 5, 2006 [Page 10] Internet-Draft SIP Connection Reuse February 2006 9. Security Considerations This document presents requirements and a mechanism for reusing existing connections easily. Unauthenticated connection reuse would present many opportunities for rampant abuse and hijacking. Authenticating connection aliases is essential to prevent connection hijacking. For example, a program run by a malicious user of a multiuser system could attempt to hijack SIP requests destined for the well-known SIP port from a large relay proxy. 9.1 Authenticating TLS Client Connections When a TLS client establishes a connection with a server, it is presented with the server's X.509 certificate. The client MUST ensure that the canonical host name of the server is present either as the distinguished name (DN) of the Subject field or as a DNS URI in the subjectAltName X.509v3 extension before updating its alias table with the resolved address. 9.2 Authenticating TLS Server Connections A TLS server conformant to this specification MUST ask for a client certificate; if the client possesses a certificate, it will be presented to the server for mutual authentication. The server MUST ensure that the canonical host name of the client is present either as the distinguished name (DN) of the Subject field or as a DNS URI in the subjectAltName X.509v3 extension before updating its alias table. If the client does not have a certificate, it is RECOMMENDED that servers issue a 403 response with the reason phrase set to "Certificate Required for Alias" to provide a more descriptive reason for rejection to a human user. The TLS connection should be closed immediately since accepting such a connection and establishing an alias would be tantamount to using an encrypted channel for TCP but still exposing the server to the same types of attacks described in Section 9.3. 9.3 Security Considerations for the TCP Transport Connection reuse over TCP is inherently insecure. Because the nature of the aliasing mechanism is such that it redirects requests destined for one port at a host to another port, service hi-jacking can result if adequate care is not taken to ensure that the redirected port is indeed authorized to receive the requests that would normally have gone to another, authorized port. Consider the following scenario to understand the service hi-jacking attack that can be mounted when using connection reuse over TCP. A TCP server receives a request with the "alias" parameter as follows Mahy, et al. Expires August 5, 2006 [Page 11] Internet-Draft SIP Connection Reuse February 2006 (the "received" parameter is added by the server after getting the request): Via: SIP/2.0/TCP uac.example.com;branch=z9hG4bKa7c8dze;alias; received=192.0.4.33 From the server's perspective, its alias table is updated such that whenever a request is destined to 192.0.4.33, port 5060, it will instead be sent to the peer at the end of the aliased connection. The security attack can now be mounted as follows: assume a malware program is running on a multi-user computer. The malware program knows that a user on the computer runs a SIP user agent, but the SIP user agent is currently not active (possibly by scanning ports on the local machine to seek a busy port 5060). Note that the malware program does not need to wait until the legitimate user agent was not running, however, doing so increases the chances that the server will not reject the malware program's request. Once the malware program decides that a legitimate user agent is not running, it sends sends a request to the server with an "alias" parameter. The server believes it is accepting a request from a legitimate user agent and sends subsequent requests to the aliased connection. The SIP service on the computer has now effectively been hi-jacked for the default port. The malware program does not need administrative privileges to execute, and in fact, can masquerade as any user (legitimate or not) of the computer. Later on, when the legitimate user agent is started, it may also send a request with an "alias" parameter to the server, which may detect that it now has two aliased connections. Making matters much worse, it cannot determine which of the two is the legitimate one and may well reject the request from the legitimate user. In another form of this attack, the legitimate user agent may not support connection aliasing, but the malware program may use the mechanism to usurp the SIP service on the computer. In yet another form of an attack, the malware program uses the aliasing mechanism to shortcut registering with a proxy to receive requests. In this case, it sends a request to the edge proxy (who may also substitute as the inbound proxy with access to a location service for that domain). In the request is a bogus request URI that will cause the edge proxy to fail the request, however, the edge proxy keeps the connection open and any subsequent requests destined to that host on the default port are instead sent to the malware program. Registration is thus not needed in order to receive incoming requests. HTTP Digest is useful to mitigate only a subset of these attacks over Mahy, et al. Expires August 5, 2006 [Page 12] Internet-Draft SIP Connection Reuse February 2006 TCP. For instance, HTTP Digest helps in authenticating a user agent to a proxy server before the alias table is updated. However, HTTP Digest is of no help when one proxy desires to enter an aliasing agreement with another downstream proxy. Keeping in view the possible attacks for TCP connection reuse documented here and the limited help provided by HTTP Digest to mitigate these attacks, it is recommended that TCP peers that want to avail of connection reuse do so such that each peer actively opens up a TCP connection in the direction of the other (as depicted in Figure 2). This manner of opening connections, while still not secure, is at least much more apparent and direct than using the connection reuse mechanism over TCP in an unauthenticated fashion. 10. Connection Reuse and SRV Interaction Connection reuse has an interaction with the DNS SRV load balancing mechanism. To understand the interaction, consider the following figure: /+---- S1 +-------+/ | Proxy |------- S2 +-------+\ \+---- S3 Figure 5: Load balancing. Here, the proxy uses DNS SRV to load balance across the three servers, S1, S2, and S3. Using the connect reuse mechanism specified in this document, over time the proxy will maintain a distinct aliased connection to each of the servers. However, once this is done, subsequent traffic is load balanced across the three downstream servers in the normal manner. 11. IANA Considerations This document adds a parameter to the SIP header field parameters registry: Header field in which parameter can appear: Via Name of the parameter: alias Reference: This document Mahy, et al. Expires August 5, 2006 [Page 13] Internet-Draft SIP Connection Reuse February 2006 12. Acknowledgments Thanks to Jon Peterson for helpful answers about certificate behavior with SIP, Jonathan Rosenberg for his initial support of this concept, and Cullen Jennings for providing a sounding board for this idea. 13. References 13.1 Normative References [1] Jennings, C. and R. Mahy, "Managing Client Initiated Connections in the Session Initiation Protocol (SIP)", draft-ietf-sip-outbound-01.txt (work in progress), October 2005. [2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [3] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", RFC 2119, March 1997. [4] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC 2246, January 1999. [5] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol (SIP): Locating SIP Servers", RFC 3263, June 2002. [6] Crocker, D. and P. Overell, "ABNF for Syntax Specifications'>Augmented BNF for Syntax Specifications: ABNF", RFC 4234, October 2005. 13.2 Informational References [7] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE Method", RFC 3311, September 2002. [8] Rosenberg, J., Peterson, J., Schulzrinne, H., and H. Camarillo, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", RFC 3725, April 2004. [9] Sparks, R., "The Session Initiation Protocol (SIP) Refer Method", RFC 3515, April 2003. [10] Roach, A., "The Session Initiation Protocol (SIP)-Specific Event Notification", RFC 3265, June 2002. [11] Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M., Zhang, L., and V. Mahy, et al. Expires August 5, 2006 [Page 14] Internet-Draft SIP Connection Reuse February 2006 Paxson, "The Session Initiation Protocol (SIP)-Specific Event Notification", RFC 2960, October 2000. Authors' Addresses Rohan Mahy SIP Edge LLC Email: rohan@ekabal.com Vijay K. Gurbani (editor) Lucent Technologies, Inc./Bell Laboratories Email: vkg at acm dot org Brett Tate BroadSoft Email: brett@broadsoft.com Mahy, et al. Expires August 5, 2006 [Page 15] Internet-Draft SIP Connection Reuse February 2006 Intellectual Property Statement The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79. Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr. The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at ietf-ipr@ietf.org. Disclaimer of Validity This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Copyright Statement Copyright (C) The Internet Society (2006). This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights. Acknowledgment Funding for the RFC Editor function is currently provided by the Internet Society. Mahy, et al. Expires August 5, 2006 [Page 16]