SIPPING Working Group C. Boulton, Ed. Internet-Draft Avaya Intended status: BCP J. Rosenberg Expires: March 21, 2009 Cisco Systems G. Camarillo Ericsson F. Audet Nortel September 17, 2008 Best Current Practices for NAT Traversal for Client-Server SIP draft-ietf-sipping-nat-scenarios-09 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 March 21, 2009. Abstract Traversal of the Session Initiation Protocol (SIP) and the sessions it establishes through Network Address Translators (NATs) is a complex problem. Currently there are many deployment scenarios and traversal mechanisms for media traffic. This document aims to provide concrete recommendations and a unified method for NAT traversal as well as documenting corresponding flows. Boulton, Ed., et al. Expires March 21, 2009 [Page 1] Internet-Draft NAT Scenarios September 2008 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 4 4. Solution Technology Outline Description . . . . . . . . . . . 7 4.1. SIP Signaling . . . . . . . . . . . . . . . . . . . . . . 8 4.1.1. Symmetric Response . . . . . . . . . . . . . . . . . . 8 4.1.2. Client Initiated Connections . . . . . . . . . . . . . 9 4.2. Media Traversal . . . . . . . . . . . . . . . . . . . . . 9 4.2.1. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . 10 4.2.2. RTCP . . . . . . . . . . . . . . . . . . . . . . . . . 10 4.2.3. ICE/STUN/TURN . . . . . . . . . . . . . . . . . . . . 10 5. NAT Traversal Scenarios . . . . . . . . . . . . . . . . . . . 12 5.1. Basic NAT SIP Signaling Traversal . . . . . . . . . . . . 12 5.1.1. Registration (Registrar/Edge Proxy Co-Located) . . . . 12 5.1.2. Registration(Registrar/Edge Proxy not Co-Located) . . 16 5.1.3. Initiating a Session . . . . . . . . . . . . . . . . . 19 5.1.4. Receiving an Invitation to a Session . . . . . . . . . 22 5.2. Basic NAT Media Traversal . . . . . . . . . . . . . . . . 26 5.2.1. Endpoint Independent NAT . . . . . . . . . . . . . . . 27 5.2.2. Address and Port Dependant NAT . . . . . . . . . . . . 46 6. IPv4-IPv6 Transition . . . . . . . . . . . . . . . . . . . . . 54 6.1. IPv4-IPv6 Transition for SIP Signaling . . . . . . . . . . 54 6.2. IPv4-IPv6 Transition for Media . . . . . . . . . . . . . . 55 7. Security Considerations . . . . . . . . . . . . . . . . . . . 56 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 57 9. IAB Considerations . . . . . . . . . . . . . . . . . . . . . . 57 10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 57 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 57 11.1. Normative References . . . . . . . . . . . . . . . . . . . 57 11.2. Informative References . . . . . . . . . . . . . . . . . . 59 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 60 Intellectual Property and Copyright Statements . . . . . . . . . . 61 Boulton, Ed., et al. Expires March 21, 2009 [Page 2] Internet-Draft NAT Scenarios September 2008 1. Introduction NAT (Network Address Translators) traversal has long been identified as a complex problem when considered in the context of the Session Initiation Protocol (SIP)[RFC3261] and it's associated media such as Real Time Protocol (RTP)[RFC3550]. The problem is exacerbated by the variety of NATs that are available in the market place today and the large number of potential deployment scenarios. Details of different NATs behavior can be found in 'NAT Behavioral Requirements for Unicast UDP' [RFC4787]. The IETF has been active on many specifications for the traversal of NATs, including STUN[I-D.ietf-behave-rfc3489bis], ICE[I-D.ietf-mmusic-ice], symmetric response[RFC3581], symmetric RTP[RFC4961], TURN[I-D.ietf-behave-turn], SIP Outbound[I-D.ietf-sip-outbound], SDP attribute for RTCP[RFC3605], and others. These each represent a part of the solution, but none of them gives the overall context for how the NATs traversal problem is decomposed and solved through this collection of specifications. This document serves to meet that need. This document provides a definitive set of 'Best Common Practices' to demonstrate the traversal of SIP and its associated media through NAT devices. The document does not propose any new functionality but does draw on existing solutions for both core SIP signaling and media traversal (as defined in Section 4). The best practices described in this document are for traditional "client- server"-style SIP. It seems likely that other groups using SIP, for example P2PSIP, will recommend these same practices between a P2PSIP client and a P2PSIP peer, but will recommend different practices for use between peers in a peer-to-peer network. The draft is split into distinct sections as follows: 1. A clear definition of the problem statement. 2. Description of proposed solutions for both SIP protocol signaling and media signaling. 3. A set of basic and advanced flow scenarios. 2. 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 [RFC2119]. Boulton, Ed., et al. Expires March 21, 2009 [Page 3] Internet-Draft NAT Scenarios September 2008 3. Problem Statement The traversal of SIP through NATs can be split into two categories that both require attention - The core SIP signaling and associated media traversal. This document assumes NATs that do not contain SIP- aware Application Layer Gateways (ALG). Some NATs that are available today contain such behavior that which makes much of the issues discussed in the document not applicable. It should also be noted that Session Border Controllers (SBC) doing 'hosted NAT traversal' also makes many of the discussions in this document moot. More information can be obtained from [I-D.ietf-sipping-sbc-funcs] and [I-D.ietf-mmusic-media-path-middleboxes]. The core SIP signaling has a number of issues when traversing through NATs. Normal SIP response routing over UDP causes the response to be delivered to the source IP address specified in the topmost Via header, or the "received" parameter of the topmost Via header. The port is extracted from the SIP 'Via' header to complete the IP address/port combination for returning the SIP response. While the destination for the response is correct, the port contained in the SIP 'Via' header represents the listening port of the originating client and not the port representing the open pin hole on the NAT. This results in responses being sent back to the NAT but to a port that is likely not open for SIP traffic. The SIP response will then be dropped at the NAT. This is illustrated in Figure 1 which depicts a SIP response being returned to port 5060. Private NAT Public Network | Network | | -------- SIP Request |open port 10923 -------- | |-------------------->--->-----------------------| | | | | | | | Client | |port 5060 SIP Response | Proxy | | | x<------------------------| | | | | | | -------- | -------- | | | Figure 1: Failed Response Boulton, Ed., et al. Expires March 21, 2009 [Page 4] Internet-Draft NAT Scenarios September 2008 Secondly, there are two cases where new requests re-use existing connections. The first is when using a reliable, connection orientated transport protocol such as TCP, SIP has an inherent mechanism that results in SIP responses reusing the connection that was created/used for the corresponding transactional request. The SIP protocol does not provide a mechanism that allows new requests generated in the reverse direction of the originating client to use, for example, the existing TCP connection created between the client and the server during registration. This results in the registered contact address not being bound to the "connection" in the case of TCP. Requests are then blocked at the NAT, as illustrated in Figure 2. The second case is when unreliable transport protocols such as UDP where external NAT mappings need to be re-used to reach a SIP entity on the private side of the network. Private NAT Public Network | Network | | -------- (UAC 8023) REGISTER/Response (UAS 5060) -------- | |-------------------->---<-----------------------| | | | | | | | Client | |5060 INVITE (UAC 8015)| Proxy | | | x<------------------------| | | | | | | -------- | -------- | | | Figure 2: Failed Request In Figure 2 the original REGISTER request is sent from the client on port 8023 and received on port 5060, establishing a connection and opening a pin-hole in the NAT. The generation of a new request from the proxy results in a request destined for the registered entity (Contact IP address) which is not reachable from the public network. This results in the new SIP request attempting to create a connection to a private network address. This problem would be solved if the original connection was re-used. While this problem has been discussed in the context of connection orientated protocols such as TCP, the problem exists for SIP signaling using any transport protocol. The impact of connection reuse of connection orientated transports (TCP, TLS, etc) is discussed in more detail in the connection reuse specification[I-D.ietf-sip-connect-reuse]. The approach proposed for this problem in Section 4 of this document is relevant for all SIP signaling in conjunction with connection reuse, Boulton, Ed., et al. Expires March 21, 2009 [Page 5] Internet-Draft NAT Scenarios September 2008 regardless of the transport protocol. NAT policy can dictate that connections should be closed after a period of inactivity. This period of inactivity may vary from a number seconds to hours. SIP signaling can not be relied upon to keep alive connections for the following two reasons. Firstly, SIP entities can sometimes have no signaling traffic for long periods of time which has the potential to exceed the inactivity timer, and this can lead to problems where endpoints are not available to receive incoming requests as the connection has been closed. Secondly, if a low inactivity timer is specified, SIP signaling is not appropriate as a keep-alive mechanism as it has the potential to add a large amount of traffic to the network which uses up valuable resource and also requires processing at a SIP stack, which is also a waste of processing resources. Media associated with SIP calls also has problems traversing NAT. RTP [RFC3550] runs over UDP and is one of the most common media transport types used in SIP signaling. Negotiation of RTP occurs with a SIP session establishment using the Session Description Protocol(SDP) [RFC4566] and a SIP offer/answer exchange[RFC3264]. During a SIP offer/answer exchange an IP address and port combination are specified by each client in a session as a means of receiving media such as RTP. The problem arises when a client advertises its address to receive media and it exists in a private network that is not accessible from outside the NAT. Figure 3 illustrates this problem. NAT Public Network NAT | | | | | | -------- | SIP Signaling Session | -------- | |----------------------->---<--------------------| | | | | | | | | Client | | | | Client | | A |>=====>RTP>==Unknown Address==>X | | B | | | | X<==Unknown Address==| | | Client |<---------------------------------send response| Client | | A | | | B | | | | | | -------- | -------- | | | Figure 4: Symmetric Response The outgoing request from Client A opens a pin hole in the NAT. Boulton, Ed., et al. Expires March 21, 2009 [Page 8] Internet-Draft NAT Scenarios September 2008 Client B would normally respond to the port available in the SIP Via header, as illustrated in Figure 1. Client B honors the 'rport' parameter in the SIP Via header and routes the response to port from which it was sent. The exact functionality for this method of response traversal is called 'Symmetric Response' and the details are documented in RFC 3581 [RFC3581]. Additional requirements are imposed on SIP entities in RFC 3581 [RFC3581] such as listening and sending SIP requests/responses from the same port. 4.1.2. Client Initiated Connections The second problem with SIP signaling, as defined in Section 3 and illustrated in Figure 2, is to allow incoming requests to be properly routed. Guidelines for devices such as User Agents that can only generate outbound connections through NATs are documented in 'Managing Client Initiated Connections in the Session Initiation Protocol(SIP)'[I-D.ietf-sip-outbound]. The document provides techniques that use a unique User Agent instance identifier (instance-id) in association with a flow identifier (reg-id). The combination of the two identifiers provides a key to a particular connection (both UDP and TCP) that is stored in association with registration bindings. On receiving an incoming request to a SIP Address-Of-Record (AOR), a proxy/registrar routes to the associated flow created by the registration and thus a route through NATs. It also provides a keepalive mechanism for clients to keep NATs bindings alive. This is achieved by multiplexing a ping/pong mechanism over the SIP signaling connection (STUN for UDP and CRLF/operating system keepalive for reliable transports like TCP). Usage of [I-D.ietf-sip-outbound] is RECOMMENDED. This mechanism is not transport specific and should be used for any transport protocol. Even if the SIP Outbound draft is not used, clients generating SIP requests SHOULD use the same IP address and port (i.e., socket) for both transmission and receipt of SIP messages. Doing so allows for the vast majority of industry provided solutions to properly function. Deployments should also consider the mechanism described in the Connection Reuse[I-D.ietf-sip-connect-reuse] specification for routing bi-directional messages securely between trusted SIP Proxy servers. 4.2. Media Traversal The issues of media traversal through NATs is not straight forward and requires the combination of a number of traversal methodologies. The technologies outlined in the remainder of this section provide the required solution set. Boulton, Ed., et al. Expires March 21, 2009 [Page 9] Internet-Draft NAT Scenarios September 2008 4.2.1. Symmetric RTP/RTCP The primary problem identified in Section 3 of this document is that internal IP address/port combinations can not be reached from the public side of NATs. In the case of media such as RTP, this will result in no audio traversing NATs (as illustrated in Figure 3). To overcome this problem, a technique called 'Symmetric RTP/ RTCP'[RFC4961] can be used. This involves a SIP endpoint both sending and receiving RTP/RTCP traffic from the same IP address/port combination. 'Symmetric RTP/RTCP' SHOULD only be used for traversal of RTP through NATs when one of the participants in a media session definitively knows that it is on the public network and is using a 'latching' technique as described in [I-D.ietf-mmusic-media-path-middleboxes]. Symmetric RTP/RTCP is important for everything that might want to traverse a NAT or speak with an endpoint that is behind a NAT - even if the remote endpoint is an SBC performing 'hosted NAT traversal'. 4.2.2. RTCP Normal practice when selecting a port for defining RTP Control Protocol (RTCP) [RFC3550] is for consecutive order numbering (i.e select an incremented port for RTCP from that used for RTP). This assumption causes RTCP traffic to break when traversing certain types of NATs due to blocked ports. To combat this problem a specific address and port need to be specified in the SDP rather than relying on such assumptions. RFC 3605 [RFC3605] defines an SDP attribute that is included to explicitly specify transport connection information for RTCP so a separate, explicit NAT binding can be set up for the purpose. The address details can be obtained using any appropriate method including those detailed in this section (e.g. STUN, TURN, ICE). The use of RFC 3605 [RFC3605] MUST be supported. An alternative mechanism defined in [I-D.ietf-avt-rtp-and-rtcp-mux] specifies 'muxing' both RTP and RTCP on the same IP/PORT combination. Using this technique eliminates the problem but is still immature. 4.2.3. ICE/STUN/TURN ICE, STUN and TURN are a suite of 3 inter-related protocols that combine to provide a complete media traversal solution for NATs. The following sections provide details of each component part. 4.2.3.1. STUN Session Traversal Utilities for NAT or STUN is defined in RFC 3489bis [I-D.ietf-behave-rfc3489bis]. STUN is a lightweight tool kit and Boulton, Ed., et al. Expires March 21, 2009 [Page 10] Internet-Draft NAT Scenarios September 2008 protocol that provides details of the external IP address/port combination used by the NAT device to represent the internal entity on the public facing side of NATs. On learning of such an external representation, a client can use it accordingly as the connection address in SDP to provide NAT traversal. Using terminology defined in the draft 'NAT Behavioral Requirements for Unicast UDP' [RFC4787], STUN does work with 'Endpoint Independent Mapping' but does not work with either 'Address Dependent Mapping' or 'Address and Port Dependent Mapping' type NATs. Using STUN with either of the previous two NATs mappings to probe for the external IP address/port representation will provide a different result to that required for traversal by an alternative SIP entity. The IP address/port combination deduced for the STUN server would be blocked for incoming packets from an alterative SIP entity. As mentioned in Section 4.1.2, STUN is also used as a client-to- server keep-alive mechanism to refresh NAT bindings. 4.2.3.2. TURN As described in the Section 4.2.3.1, the STUN protocol does not work for UDP traversal through certain identified NAT mappings. 'Traversal Using Relays around NAT' is a usage of the STUN protocol for deriving (from a TURN server) an address that will be used to relay packets towards a client. TURN provides an external address (globally routable) at a STUN server that will act as a media relay which attempts to allow traffic to reach the associated internal address. The full details of the TURN specification are defined in [I-D.ietf-behave-turn]. A TURN service will almost always provide media traffic to a SIP entity but it is RECOMMENDED that this method would only be used as a last resort and not as a general mechanism for NAT traversal. This is because using TURN has high performance costs when relaying media traffic and can lead to unwanted latency. 4.2.3.3. ICE Interactive Connectivity Establishment (ICE) is the RECOMMENDED method for traversal of existing NATs if Symmetric RTP is not appropriate. ICE is a methodology for using existing technologies such as STUN, TURN and any other UNSAF[RFC3424] compliant protocol to provide a unified solution. This is achieved by obtaining as many representative IP address/port combinations as possible using technologies such as STUN/TURN (*note - an ICE endpoint can also use non-IETF mechanisms (e.g., NAT-PMP, UPnP IGD) to learn public IP addresses and ports, and populate a=candidate lines with that information). Once the addresses are accumulated, they are all included in the SDP exchange in a new media attribute called 'candidate'. Each 'candidate' SDP attribute entry has detailed Boulton, Ed., et al. Expires March 21, 2009 [Page 11] Internet-Draft NAT Scenarios September 2008 connection information including a media address, priority and transport protocol. The appropriate IP address/port combinations are used in the order specified by the priority. A client compliant to the ICE specification will then locally run STUN servers on all addresses being advertised using ICE. Each instance will undertake connectivity checks to ensure that a client can successfully receive media on the advertised address. Only connections that pass the relevant connectivity checks are used for media exchange. The full details of the ICE methodology are contained in [I-D.ietf-mmusic-ice]. 5. NAT Traversal Scenarios This section of the document includes detailed NAT traversal scenarios for both SIP signaling and the associated media. Signalling NAT traversal is achieved using [I-D.ietf-sip-outbound]. 5.1. Basic NAT SIP Signaling Traversal The following sub-sections concentrate on SIP signaling traversal of NATs. The scenarios include traversal for both reliable and un- reliable transport protocols. 5.1.1. Registration (Registrar/Edge Proxy Co-Located) The set of scenarios in this section document basic signaling traversal of a SIP REGISTER method through NATs. 5.1.1.1. UDP Boulton, Ed., et al. Expires March 21, 2009 [Page 12] Internet-Draft NAT Scenarios September 2008 Registrar/ Bob NAT Edge Proxy | | | |(1) REGISTER | | |----------------->| | | | | | |(1) REGISTER | | |----------------->| | | | |*************************************| | Create Outbound Connection Tuple | |*************************************| | | | | |(2) 200 OK | | |<-----------------| | | | |(2) 200 OK | | |<-----------------| | | | | Figure 5: UDP Registration In this example the client sends a SIP REGISTER request through a NAT. The client will include an 'rport' parameter as described in Section 4.1.1 of this document for allowing traversal of UDP responses. The original request as illustrated in (1) in Figure 5 is a standard REGISTER message: Message 1: REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob ;tag=7F94778B653B To: Bob Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Contact: ;reg-id=1 ;+sip.instance="" Content-Length: 0 This SIP transaction now generates a SIP 200 OK response, as depicted in (2) from Figure 5: Message 2: Boulton, Ed., et al. Expires March 21, 2009 [Page 13] Internet-Draft NAT Scenarios September 2008 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2;rport=8050;branch=z9hG4bKnashds7; received=172.16.3.4 From: Bob ;tag=7F94778B653B To: Bob ;tag=6AF99445E44A Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Require: outbound Contact: ;reg-id=1;expires=3600 ;+sip.instance="" Content-Length: 0 The response will be sent to the address appearing in the 'received' parameter of the SIP 'Via' header (address 172.16.3.4). The response will not be sent to the port deduced from the SIP 'Via' header, as per standard SIP operation but will be sent to the value that has been stamped in the 'rport' parameter of the SIP 'Via' header (port 8050). For the response to successfully traverse the NAT, all of the conventions defined in RFC 3581 [RFC3581] MUST be obeyed. Make note of both the 'reg-id' and 'sip.instance' contact header parameters. They are used to establish an Outbound connection tuple as defined in [I-D.ietf-sip-outbound]. The connection tuple creation is clearly shown in Figure 5. This ensures that any inbound request that causes a registration lookup will result in the re-use of the connection path established by the registration. This exonerates the need to manipulate contact header URIs to represent a globally routable address as perceived on the public side of a NAT. 5.1.1.2. Connection Oriented Transport Boulton, Ed., et al. Expires March 21, 2009 [Page 14] Internet-Draft NAT Scenarios September 2008 Registrar/ Bob NAT Edge Proxy | | | |(1) REGISTER | | |----------------->| | | | | | |(1) REGISTER | | |----------------->| | | | |*************************************| | Create Outbound Connection Tuple | |*************************************| | | | | |(2) 200 OK | | |<-----------------| | | | |(2) 200 OK | | |<-----------------| | | | | Figure 6 Traversal of SIP REGISTER requests/responses using a reliable, connection orientated protocol such as TCP does not require any additional core SIP signaling extensions, beyond the procedures defined in [I-D.ietf-sip-outbound]. SIP responses will re-use the connection created for the initial REGISTER request, (1) from Figure 6: Message 1: REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob ;tag=7F94778B653B To: Bob Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Contact: ;reg-id=1 ;+sip.instance="" Content-Length: 0 Message 2: Boulton, Ed., et al. Expires March 21, 2009 [Page 15] Internet-Draft NAT Scenarios September 2008 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7 From: Bob ;tag=7F94778B653B To: Bob ;tag=6AF99445E44A Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Require: outbound Contact: ;reg-id=1;expires=3600 ;+sip.instance="" Content-Length: 0 This example was included to show the inclusion of the +sip.instance Contact header parameter as defined in the SIP Outbound specification [I-D.ietf-sip-outbound]. This creates an association tuple as described in the previous example for future inbound requests directed at the newly created registration binding with the only difference that the association is with a TCP connection, not a UDP pin hole binding. 5.1.2. Registration(Registrar/Edge Proxy not Co-Located) This section demonstrates traversal mechanisms when the Registrar component is not co-located with the edge proxy element. The procedures described in this section are identical, regardless of transport protocol and so only one example will be documented in the form of TCP. Boulton, Ed., et al. Expires March 21, 2009 [Page 16] Internet-Draft NAT Scenarios September 2008 Bob NAT Edge Proxy Registrar | | | | |(1) REGISTER | | | |----------------->| | | | | | | | |(1) REGISTER | | | |----------------->| | | | | | | | |(2) REGISTER | | | |----------------->| | | | | |********************************************************| | Create Outbound Connection Tuple | |********************************************************| | | | | | | |(3) 200 OK | | | |<-----------------| | |(4)200 OK | | | |<-----------------| | | | | | |(4)200 OK | | | |<-----------------| | | | | | | Figure 7: Registration(Registrar/Proxy not Co-Located) This scenario builds on the previous example contained in Section 5.1.1.2. The primary difference being that the REGISTER request is routed onwards from a Proxy Server to a separated Registrar. The important message to note is (1) in Figure 7. The Edge proxy, on receiving a REGISTER request that contains a 'sip.instance' media feature tag, forms a unique flow identifier token as discussed in [I-D.ietf-sip-outbound]. At this point, the proxy server routes the SIP REGISTER message to the Registrar. The proxy will create the connection tuple as described in SIP Outbound at the same moment as the co-located example, but for subsequent messages to arrive at the Proxy, the proxy needs to indicate its need to remain in the SIP signaling path. To achieve this the proxy inserts to REGISTER message (2) a SIP PATH extension header, as defined in RFC 3327 [RFC3327]. The previously created flow association token is inserted in a position within the Path header where it can easily be retrieved at a later point when receiving messages to be routed to the registration binding (in this case the user part of the SIP URI). The REGISTER message of (1) includes a SIP Route header for the edge proxy. Message 1: Boulton, Ed., et al. Expires March 21, 2009 [Page 17] Internet-Draft NAT Scenarios September 2008 REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob ;tag=7F94778B653B To: Bob Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Route: Contact: ;reg-id=1 ;+sip.instance="" Content-Length: 0 When proxied in (2) looks as follows: Message 2: REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/TCP ep1.example.com;branch=z9hG4bKnuiqisi Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7 Max-Forwards: 69 From: Bob ;tag=7F94778B653B To: Bob Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Contact: ;reg-id=1 ;+sip.instance="" Path: Content-Length: 0 This REGISTER request results in the Path header being stored along with the AOR and it's associated binding at the Registrar. The URI contained in the Path header will be inserted as a pre-loaded SIP 'Route' header into any request that arrives at the Registrar and is directed towards the associated AOR binding. This all but guarantees that all requests for the new registration will be forwarded to the Edge Proxy. In our example, the user part of the SIP 'Path' header URI that was inserted by the Edge Proxy contains the unique token identifying the flow to the client. On receiving subsequent requests, the edge proxy will examine the user part of the pre-loaded SIP 'route' header and extract the unique flow token for use in its connection tuple comparison, as defined in the SIP Outbound specification [I-D.ietf-sip-outbound]. An example which builds on this scenario (showing an inbound request to the AOR) is detailed in Section 5.1.4.2 of this document. Boulton, Ed., et al. Expires March 21, 2009 [Page 18] Internet-Draft NAT Scenarios September 2008 5.1.3. Initiating a Session This section covers basic SIP signaling when initiating a call from behind a NAT. 5.1.3.1. UDP Initiating a call using UDP (the Edge Proxy and Authoritative Proxy funcationality are co-located). Boulton, Ed., et al. Expires March 21, 2009 [Page 19] Internet-Draft NAT Scenarios September 2008 Edge Proxy/ Bob NAT Auth. Proxy Alice | | | | |(1) INVITE | | | |----------------->| | | | | | | | |(1) INVITE | | | |----------------->| | | | | | | | |(2) INVITE | | | |---------------->| | | | | | | |(3)180 RINGING | | | |<----------------| | | | | | |(4)180 RINGING | | | |<-----------------| | | | | | |(4)180 RINGING | | | |<-----------------| | | | | | | | | |(5)200 OK | | | |<----------------| | | | | | |(6)200 OK | | | |<-----------------| | | | | | |(6)200 OK | | | |<-----------------| | | | | | | |(7)ACK | | | |----------------->| | | | | | | | |(7)ACK | | | |----------------->| | | | | | | | |(8) ACK | | | |---------------->| | | | | Figure 8: Initiating a Session - UDP The initiating client generates an INVITE request that is to be sent through the NAT to a Proxy server. The INVITE message is represented in Figure 8 by (1) and is as follows: Message 1: Boulton, Ed., et al. Expires March 21, 2009 [Page 20] Internet-Draft NAT Scenarios September 2008 INVITE sip:alice@a.example SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob ;tag=ldw22z To: Alice Call-ID: 95KGsk2V/Eis9LcpBYy3 CSeq: 1 INVITE Supported: outbound Route: Contact: Content-Type: application/sdp Content-Length: ... [SDP not shown] There are a number of points to note with this message: 1. Firstly, as with the registration example in Section 5.1.1.1, responses to this request will not automatically pass back through a NAT and so the SIP 'Via' header 'rport' is included as described in the 'Symmetric response' Section 4.1.1 and defined in RFC 3581 [RFC3581]. 2. Secondly, the contact inserted contains to ensure that all new requests will be sent to the same flow. Alternatively, a GRUU might have been used. See 4.3/[I-D.ietf-sip-outbound]. In (2), the proxy inserts itself in the Via, adds the rport port number in the previous Via header, adds the received parameter in the previous Via, removes the Route header, and inserts a Record-Route with a token. Message 2: INVITE sip:alice@172.16.1.4 SIP/2.0 Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4bKnuiqisi Via: SIP/2.0/UDP 192.168.1.2;rport=8050;branch=z9hG4bKnashds7; received=172.16.3.4 Max-Forwards: 69 From: Bob ;tag=ldw22z To: Alice Call-ID: 95KGsk2V/Eis9LcpBYy3 CSeq: 1 INVITE Supported: outbound Record-Route: Contact: Content-Type: application/sdp Content-Length: ... Boulton, Ed., et al. Expires March 21, 2009 [Page 21] Internet-Draft NAT Scenarios September 2008 [SDP not shown] 5.1.3.2. Connection-oriented Transport When using a reliable transport such as TCP the call flow and procedures for traversing a NAT are almost identical to those described in Section 5.1.3.1. The primary difference when using reliable transport protocols is that Symmetric response[RFC3581] are not required for SIP responses to traverse a NAT. RFC 3261[RFC3261] defines procedures for SIP response messages to be sent back on the same connection on which the request arrived. See section 9.5/ [I-D.ietf-sip-outbound] for an example call flow of an outgoing call. 5.1.4. Receiving an Invitation to a Session This section details scenarios where a client behind a NAT receives an inbound request through a NAT. These scenarios build on the previous registration scenario from Section 5.1.1 and Section 5.1.2 in this document. 5.1.4.1. Registrar/Proxy Co-located The SIP signaling on the interior of the network (behind the user's proxy) is not impacted directly by the transport protocol and so only one example scenario is necessary. The example uses UDP and follows on from the registration installed in the example from Section 5.1.1.1. Edge Proxy Bob NAT Auth. Proxy Alice | | | | |*******************************************************| | Registration Binding Installed in | | section 5.1.1.1 | |*******************************************************| | | | | | | |(1)INVITE | | | |<----------------| | | | | | |(2)INVITE | | | |<-----------------| | | | | | |(2)INVITE | | | |<-----------------| | | | | | | | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 22] Internet-Draft NAT Scenarios September 2008 Figure 9: Receiving an Invitation to a Session An INVITE request arrives at the Authoritative Proxy with a destination pointing to the AOR of that inserted in Section 5.1.1.1. The message is illustrated by (1) in Figure 9 and looks as follows: INVITE sip:bob@example.com SIP/2.0 Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d Max-Forwards: 70 From: External Alice ;tag=02935 To: Bob Call-ID: klmvCxVWGp6MxJp2T2mb CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: .. [SDP not shown] The INVITE request matches the registration binding previously installed at the Registrar and the INVITE request-URI is re-written to the selected onward address. The proxy then examines the request URI of the INVITE and compares with its list of current open flows. It uses the incoming AOR to commence the check for associated open connections/mappings. Once matched, the proxy checks to see if the unique instance identifier (+sip.instance) associated with the binding equals the same instance identifier associated with the flow. The request is then dispatched on the appropriate flow. This is message (2) from Figure 9 and is as follows: INVITE sip:bob@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4kmlds893jhsd Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d Max-Forwards: 69 From: Alice ;tag=02935 To: client bob Call-ID: klmvCxVWGp6MxJp2T2mb CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: .. [SDP not shown] It is a standard SIP INVITE request with no additional functionality. Boulton, Ed., et al. Expires March 21, 2009 [Page 23] Internet-Draft NAT Scenarios September 2008 The major difference being that this request will not be forwarded to the address specified in the Request-URI, as standard SIP rules would enforce but will be sent on the flow associated with the registration binding (look-up procedures in RFC 3263 [RFC3263] are overridden). This then allows the original connection/mapping from the initial registration process to be re-used. 5.1.4.2. Edge Proxy/Authoritative Proxy Not Co-located The core SIP signaling associated with this call flow is not impacted directly by the transport protocol and so only one example scenario is necessary. The example uses UDP and follows on from the registration installed in the example from Section 5.1.2. Bob NAT Edge Proxy Auth. Proxy Alice | | | | | |***********************************************************| | Registration Binding Installed in | | section 5.1.2 | |***********************************************************| | | | | | | | | |(1)INVITE | | | | |<-------------| | | | | | | | |(2)INVITE | | | | |<-------------| | | | | | | | |(3)INVITE | | | | |<-------------| | | | | | | | |(3)INVITE | | | | |<-------------| | | | | | | | | | | | | | Figure 10: Registrar/Proxy Not Co-located An INVITE request arrives at the Authoritative Proxy with a destination pointing to the AOR of that inserted in Section 5.1.2. The message is illustrated by (1) in Figure 10 and looks as follows: Boulton, Ed., et al. Expires March 21, 2009 [Page 24] Internet-Draft NAT Scenarios September 2008 INVITE sip:bob@example.com SIP/2.0 Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d Max-Forwards: 70 From: Alice ;tag=02935 To: Bob Call-ID: klmvCxVWGp6MxJp2T2mb CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: .. [SDP not shown] The INVITE request matches the registration binding previously installed at the Registrar and the INVITE request-URI is re-written to the selected onward address. The Registrar also identifies that a SIP PATH header was associated with the registration and pushes it into the INVITE request in the form of a pre-loaded SIP Route header. It then forwards the request on to the proxy identified in the SIP Route header as shown in (2) from Figure 10: INVITE sip:bob@client.example.com SIP/2.0 Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bK74fmljnc Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d Route: Max-Forwards: 69 From: Alice ;tag=02935 To: Bob Call-ID: klmvCxVWGp6MxJp2T2mb CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: .. [SDP not shown] The request then arrives at the outbound proxy for the client. The proxy examines the request URI of the INVITE in conjunction with the flow token that it previously inserted into the user part of the PATH header SIP URI (which now appears in the user part of the Route header in the incoming INVITE). The proxy locates the appropriate flow and sends the message to the client, as shown in (3) from Figure 10: Boulton, Ed., et al. Expires March 21, 2009 [Page 25] Internet-Draft NAT Scenarios September 2008 INVITE sip:bob@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4nsi30dncmnl Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bK74fmljnc Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d Record-Route: Max-Forwards: 68 From: Alice ;tag=02935 To: bob Call-ID: klmvCxVWGp6MxJp2T2mb CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: .. [SDP not shown] It is a standard SIP INVITE request with no additional functionality at the originator. The major difference being that this request will not follow the address specified in the Request-URI when it reaches the outbound proxy, as standard SIP rules would enforce but will be sent on the flow associated with the registration binding as indicated in the Route header(look-up procedures in RFC 3263 [RFC3263] are overridden). This then allows the original connection/ mapping from the initial registration to the outbound proxy to be re- used. 5.2. Basic NAT Media Traversal This section provides example scenarios to demonstrate basic media traversal using the techniques outlined earlier in this document. In the flow diagrams STUN messages have been annotated for simplicity as follows: o The "Src" attribute represents the source transport address of the message. o The "Dest" attribute represents the destination transport address of the message. o The "Map" attribute represents the server reflexive (XOR-MAPPED- ADDRESS STUN attribute) transport address. o The "Rel" attribute represents the relayed (RELAY-ADDRESS STUN attribute) transport address. The meaning of each STUN attribute is extensively explained in the core STUN[I-D.ietf-behave-rfc3489bis] and TURN [I-D.ietf-behave-turn] drafts. A number of ICE SDP attributes have also been included in some of the examples. Detailed information on individual attributes can be Boulton, Ed., et al. Expires March 21, 2009 [Page 26] Internet-Draft NAT Scenarios September 2008 obtained from the core ICE specification[I-D.ietf-mmusic-ice]. The examples also contain a mechanism for representing transport addresses. It would be confusing to include representations of network addresses in the call flows and make them hard to follow. For this reason network addresses will be represented using the following annotation. The first component will contain the representation of the client responsible for the address. For example in the majority of the examples "L" (left client), "R" (right client), NAT-PUB" (NAT public), PRIV (Private), and "STUN-PUB" (STUN Public) are used. To allow for multiple addresses from the same network element, each representation can also be followed by a number. These can also be used in combination. For example "L-NAT- PUB-1" would represent a public network address of the left hand side NAT while "R-NAT-PUB-1" would represent a public network address of the right hand side of the NAT. "L-PRIV-1" would represent a private network address of the left hand side of the NAT while "R-PRIV-1" represents a private address of the right hand side of the NAT. It should also be noted that during the examples it might be appropriate to signify an explicit part of a transport address. This is achieved by adding either the '.address' or '.port' tag on the end of the representation. For example, 'L-PRIV-1.address' and 'L-PRIV- 1.port'. The use of '$' signifies variable parts in example SIP messages. 5.2.1. Endpoint Independent NAT This section demonstrates an example of a client both initiating and receiving calls behind an 'Endpoint independent' NAT. An example is included for both STUN and ICE with ICE being the RECOMMENDED mechanism for media traversal. 5.2.1.1. STUN Solution It is possible to traverse media through an 'Endpoint Independent NAT using STUN. The remainder of this section provides simplified examples of the 'Binding Discovery' STUN as defined in [I-D.ietf-behave-rfc3489bis]. The STUN messages have been simplified and do not include 'Shared Secret' requests used to obtain the temporary username and password. 5.2.1.1.1. Initiating Session The following example demonstrates media traversal through a NAT with 'Address-Independent' properties using the STUN 'Binding Discovery' usage. It is assumed in this example that the STUN client and SIP Boulton, Ed., et al. Expires March 21, 2009 [Page 27] Internet-Draft NAT Scenarios September 2008 Client are co-located on the same physical machine. Note that some SIP signaling messages have been left out for simplicity. Client NAT STUN [..] Server | | | | |(1) BIND Req | | | |Src=L-PRIV-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | | |(2) BIND Req | | | |Src=NAT-PUB-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | |(3) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=NAT-PUB-1 | | | |Map=NAT-PUB-1 | | | | | | |(4) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=L-PRIV-1 | | | |Map=NAT-PUB-1 | | | | | | | |(5) BIND Req | | | |Src=L-PRIV-2 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | | |(6) BIND Req | | | |Src=NAT-PUB-2 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | |(7) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=NAT-PUB-2 | | | |Map=NAT-PUB-2 | | | | | | |(8) BIND Resp | | | |<-----------------| | | Boulton, Ed., et al. Expires March 21, 2009 [Page 28] Internet-Draft NAT Scenarios September 2008 |Src=STUN-PUB | | | |Dest=L-PRIV-2 | | | |Map=NAT-PUB-2 | | | | | | | |(9)SIP INVITE | | | |----------------->| | | | | | | | |(10)SIP INVITE | | | |------------------------------------>| | | | | | | |(11)SIP 200 OK | | |<------------------------------------| | | | | |(12)SIP 200 OK | | | |<-----------------| | | | | | | |========================================================| |>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>>| |========================================================| | | |========================================================| |<<<<<<<<<<<>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>| |========================================================| | | |========================================================| |<<<<<<<<<<<| | | | | | | | |(14) SIP ACK | | | |------------------------------------>| | | | | Figure 11: Endpoint Independent NAT - Initiating o On deciding to initiate a SIP voice session the client starts a local STUN client on the interface and port that is to be used for media (send/receive). The STUN client generates a standard 'Binding Discovery' request as indicated in (1) from Figure 11 which also highlights the source address and port for which the client device wishes to obtain a mapping. The 'Binding Discovery' Boulton, Ed., et al. Expires March 21, 2009 [Page 29] Internet-Draft NAT Scenarios September 2008 request is sent through the NAT towards the public internet and STUN server. o Message (2) traverses the NAT and breaks out onto the public internet towards the public STUN server. Note that the source address of the 'Binding Discovery' request now represents the public address and port from the public side of the NAT. o The STUN server receives the request and processes it appropriately. This results in a successful 'Binding Discovery' response being generated and returned (3). The message contains details of the XOR mapped public address (contained in the STUN XOR-MAPPED-ADDRESS attribute) which is to be used by the originating client to receive media (see 'Map=NAT-PUB-1' from (3)). o The 'Binding Discovery' response traverses back through the NAT using the path created by the 'Binding Discovery' request and presents the new XOR mapped address to the client (4). At this point the process is repeated to obtain a second XOR-mapped address (as shown in (5)-(8)) for a second local address (Address has changed from "L-PRIV-1" to "L-PRIV-2") for an RTCP port. o The client now constructs a SIP INVITE message(9). Note that traversal of SIP is not covered in this example and is discussed in Section 5.1. The INVITE request will use the addresses it has obtained in the previous STUN transactions to populate the SDP of the SIP INVITE as shown below: v=0 o=test 2890844526 2890842807 IN IP4 $L-PRIV-1.address c=IN IP4 $NAT-PUB-1.address t=0 0 m=audio $NAT-PUB-1.port RTP/AVP 0 a=rtcp:$NAT-PUB-2.port o Note that the XOR-mapped address obtained from the 'Binding Discovery' transactions are inserted as the connection address for the SDP (c=$NAT-PUB-1.address). The Primary port for RTP is also inserted in the SDP (m=audio $NAT-PUB-1.port RTP/AVP 0). Finally, the port gained from the additional 'Binding Discovery' is placed in the RTCP attribute (as discussed in Section 4.2.2) for traversal of RTCP (a=rtcp:$NAT-PUB-2.port). o The SIP signaling then traverses the NAT and sets up the SIP session (9-12). Note that the left client transmits media as soon as the 200 OK to the INVITE arrives at the client (12). Up until this point the incoming media and RTCP to the left hand client will not pass through the NAT as no outbound association has been created with the far end client. Two way media communication has now been established. Boulton, Ed., et al. Expires March 21, 2009 [Page 30] Internet-Draft NAT Scenarios September 2008 5.2.1.1.2. Receiving Session Invitation Receiving a session for an 'Endpoint Independent' NAT using the STUN 'Binding Discovery' usage is very similar to the example outlined in Section 5.2.1.1.1. Figure 12 illustrates the associated flow of messages. Client NAT STUN [..] Server | | | (1)SIP INVITE | | |<------------------------------------| | | | | |(2) SIP INVITE | | | |<-----------------| | | | | | | |(3) BIND Req | | | |Src=L-PRIV-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | | |(4) BIND Req | | | |Src=NAT-PUB-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | |(5) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=NAT-PUB-1 | | | |Map=NAT-PUB-1 | | | | | | |(6) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=L-PRIV-1 | | | |Map=NAT-PUB-1 | | | | | | | |(7) BIND Req | | | |Src=L-PRIV-2 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | | |(8) BIND Req | | | |Src=NAT-PUB-2 | | | |Dest=STUN-PUB | | | |----------------->| | Boulton, Ed., et al. Expires March 21, 2009 [Page 31] Internet-Draft NAT Scenarios September 2008 | | | | | |(9) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=NAT-PUB-2 | | | |Map=NAT-PUB-2 | | | | | | |(10) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=L-PRIV-2 | | | |Map=NAT-PUB-2 | | | | | | | |(11)SIP 200 OK | | | |----------------->| | | | |(12)SIP 200 OK | | | |------------------------------------>| | | | | |========================================================| |>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>>| |========================================================| | | | | |========================================================| |<<<<<<<<<<<<>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>| |========================================================| | | | | |========================================================| |<<<<<<<<<<<<| | | | | | | | | | |(2) Alloc Req | | | | |Src=L-NAT-PUB-1 | | | | |Dest=TURN-PUB-1 | | | | |--------------->| | | | | | | | | |(3) Alloc Resp | | | | |<---------------| | | | |Src=TURN-PUB-1 | | | | |Dest=L-NAT-PUB-1| | | | |Map=L-NAT-PUB-1 | | | | |Rel=TURN-PUB-2 | | | | | | | | |(4) Alloc Resp | | | | |<---------------| | | | |Src=TURN-PUB-1 | | | | |Dest=L-PRIV-1 | | | | |Map=L-NAT-PUB-1 | | | | |Rel=TURN-PUB-2 | | | | | | | | | |(5) Alloc Req | | | | |Src=L-PRIV-2 | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 34] Internet-Draft NAT Scenarios September 2008 |Dest=TURN-PUB-1 | | | | |--------------->| | | | | | | | | | |(6) Alloc Req | | | | |Src=L-NAT-PUB-2 | | | | |Dest=TURN-PUB-1 | | | | |--------------->| | | | | | | | | |(7) Alloc Resp | | | | |<---------------| | | | |Src=TURN-PUB-1 | | | | |Dest=NAT-PUB-2 | | | | |Map=NAT-PUB-2 | | | | |Rel=TURN-PUB-3 | | | | | | | | |(8) Alloc Resp | | | | |<---------------| | | | |Src=TURN-PUB-1 | | | | |Dest=L-PRIV-2 | | | | |Map=L-NAT-PUB-2 | | | | |Rel=TURN-PUB-3 | | | | | | | | | |(9) SIP INVITE | | | | |------------------------------------------------->| | | | | | | | | | |(10) SIP INVITE | | | | |--------------->| | | | | | | | | |(11) Alloc Req | | | | |<---------------| | | | |Src=R-PRIV-1 | | | | |Dest=TURN-PUB-1 | | | | | | | | |(12) Alloc Req | | | | |<---------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=TURN-PUB-1 | | | | | | | | | |(13) Alloc Res | | | | |--------------->| | | | |Src=TURN-PUB-1 | | | | |Dest=R-NAT-PUB-1| | | | |Map=R-NAT-PUB-1 | | | | |Rel=TURN-PUB-4 | | | | | | | | | | |(14) Alloc Res | | | | |--------------->| | | | |Src=TURN-PUB-1 | Boulton, Ed., et al. Expires March 21, 2009 [Page 35] Internet-Draft NAT Scenarios September 2008 | | | |Dest=R-PRIV-1 | | | | |Map=R-NAT-PUB-1 | | | | |Rel=TURN-PUB-4 | | | | | | | | | |(15) Alloc Req | | | | |<---------------| | | | |Src=R-PRIV-2 | | | | |Dest=TURN-PUB-1 | | | | | | | | |(16) Alloc Req | | | | |<---------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=TURN-PUB-1 | | | | | | | | | |(17) Alloc Res | | | | |--------------->| | | | |Src=TURN-PUB-1 | | | | |Dest=R-NAT-PUB-2| | | | |Map=R-NAT-PUB-2 | | | | |Rel=TURN-PUB-5 | | | | | | | | | | |(18) Alloc Res | | | | |--------------->| | | | |Src=TURN-PUB-1 | | | | |Dest=R-PRIV-2 | | | | |Map=R-NAT-PUB-2 | | | | |Rel=TURN-PUB-5 | | | | | | | | | |(19) SIP 200 OK | | |<-------------------------------------------------| | | | | | |(20) SIP 200 OK | | | | |<---------------| | | | | | | | | |(21) SIP ACK | | | | |------------------------------------------------->| | | | | | | | | | |(22) SIP ACK | | | | |--------------->| | | | | | |(23) Bind Req | | | | |------------------------>x | | | |Src=L-PRIV-1 | | | | |Dest=R-PRIV-1 | | | | | | | | | |(24) Bind Req | | | | |--------------->| | | | |Src=L-PRIV-1 | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 36] Internet-Draft NAT Scenarios September 2008 |Dest=R-NAT-PUB-1| | | | | | | | | | |(25) Bind Req | | | | |-------------------------------->| | | |Src=L-NAT-PUB-1 | | | | |Dest=R-NAT-PUB-1| | | | | | | | | | | |(26) Bind Req | | | | |--------------->| | | | |Src=L-NAT-PUB-1 | | | | |Dest=R-PRIV-1 | | | | | | | | | |(27) Bind Res | | | | |<---------------| | | | |Src=R-PRIV-1 | | | | |Dest=L-NAT-PUB-1| | | | |Map=L-NAT-PUB-1 | | | | | | | | |(28) Bind Res | | | |<--------------------------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-NAT-PUB-1| | | | |Map=L-NAT-PUB-1 | | | | | | | |(29) Bind Res | | | | |<---------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-PRIV-1 | | | | |Map=L-NAT-PUB-1 | | | | | | | | | |===================================================================| |>>>>>>>>>>>>>>>>>>Outgoing RTP sent from L-PRIV-1 >>>>>>>>>>>>>>>>>| |===================================================================| | | | | | | | | |(30) Bind Req | | | | x<-----------------------| | | | |Src=R-PRIV-1 | | | | |Dest=L-PRIV-1 | | | | | | | | | |(31) Bind Req | | | | |<---------------| | | | |Src=R-PRIV-1 | | | | |Dest=L-NAT-PUB-1| | | | | | | | |(32) Bind Req | | | |<--------------------------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-NAT-PUB-1| | Boulton, Ed., et al. Expires March 21, 2009 [Page 37] Internet-Draft NAT Scenarios September 2008 | | | | | |(33) Bind Req | | | | |<---------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-PRIV-1 | | | | | | | | | |(34) Bind Res | | | | |--------------->| | | | |Src=L-PRIV-1 | | | | |Dest=R-NAT-PUB-1| | | | |Map=R-NAT-PUB-1 | | | | | | | | | | |(35) Bind Res | | | | |-------------------------------->| | | |Src=L-NAT-PUB-1 | | | | |Dest=R-NAT-PUB-1| | | | |Map=R-NAT-PUB-1 | | | | | | | | | | | |(36) Bind Res | | | | |--------------->| | | | |Src=L-NAT-PUB-1 | | | | |Dest=R-PRIV-1 | | | | |Map=R-NAT-PUB-1 | | | | | | |===================================================================| |<<<<<<<<<<<<<<<<<| | | | |Src=L-PRIV-1 | | | | |Dest=R-NAT-PUB-1| | | | |USE-CANDIDATE | | | | | | | | | | |(38) Bind Req | | | | |-------------------------------->| | | |Src=L-NAT-PUB-1 | | | | |Dest=R-NAT-PUB-1| | | | |USE-CANDIDATE | | | | | | | | | | | |(39) Bind Req | | | | |--------------->| | | | |Src=L-NAT-PUB-1 | | | | |Dest=R-PRIV-1 | | | | |USE-CANDIDATE | | | | | | | | | |(40) Bind Res | | | | |<---------------| | | | |Src=R-PRIV-1 | Boulton, Ed., et al. Expires March 21, 2009 [Page 38] Internet-Draft NAT Scenarios September 2008 | | | |Dest=L-NAT-PUB-1| | | | |Map=L-NAT-PUB-1 | | | | | | | | |(41) Bind Res | | | |<--------------------------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-NAT-PUB-1| | | | |Map=L-NAT-PUB-1 | | | | | | | |(42) Bind Res | | | | |<---------------| | | | |Src=R-NAT-PUB-1 | | | | |Dest=L-PRIV-1 | | | | |Map=L-NAT-PUB-1 | | | | | | | | | |(43) Bind Req | | | | |--------------->| | | | |Src=L-PRIV-2 | | | | |Dest=R-NAT-PUB-2| | | | | | | | | | |(44) Bind Req | | | | |-------------------------------->| | | |Src=L-NAT-PUB-2 | | | | |Dest=R-NAT-PUB-2| | | | | | | | | | | |(45) Bind Req | | | | |--------------->| | | | |Src=L-NAT-PUB-2 | | | | |Dest=R-PRIV-2 | | | | | | | | | |(46) Bind Res | | | | |<---------------| | | | |Src=R-PRIV-2 | | | | |Dest=L-NAT-PUB-2| | | | |Map=L-NAT-PUB-2 | | | | | | | | |(47) Bind Res | | | |<--------------------------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-NAT-PUB-2| | | | |Map=L-NAT-PUB-2 | | | | | | | |(48) Bind Res | | | | |<---------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-PRIV-2 | | | | |Map=L-NAT-PUB-2 | | | | | | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 39] Internet-Draft NAT Scenarios September 2008 |===================================================================| |>>>>>>>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2 >>>>>>>>>>>>>>>>| |===================================================================| | | | | | | | | |(49) Bind Req | | | | |<---------------| | | | |Src=R-PRIV-2 | | | | |Dest=L-NAT-PUB-2| | | | | | | | |(50) Bind Req | | | |<--------------------------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-NAT-PUB-2| | | | | | | |(51) Bind Req | | | | |<---------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-PRIV-2 | | | | | | | | | |(52) Bind Res | | | | |--------------->| | | | |Src=L-PRIV-2 | | | | |Dest=R-NAT-PUB-2| | | | |Map=R-NAT-PUB-2 | | | | | | | | | | |(53) Bind Res | | | | |-------------------------------->| | | |Src=L-NAT-PUB-2 | | | | |Dest=R-NAT-PUB-2| | | | |Map=R-NAT-PUB-2 | | | | | | | | | | | |(54) Bind Res | | | | |--------------->| | | | |Src=L-NAT-PUB-2 | | | | |Dest=R-PRIV-2 | | | | |Map=R-NAT-PUB-2 | | | | | | |===================================================================| |<<<<<<<<<<<<<<<<<| | | | |Src=L-PRIV-2 | | | | |Dest=R-NAT-PUB-2| | | | |USE-CANDIDATE | | | | | | | | | | |(56) Bind Req | | | | |-------------------------------->| | Boulton, Ed., et al. Expires March 21, 2009 [Page 40] Internet-Draft NAT Scenarios September 2008 | |Src=L-NAT-PUB-2 | | | | |Dest=R-NAT-PUB-2| | | | |USE-CANDIDATE | | | | | | | | | | | |(57) Bind Req | | | | |--------------->| | | | |Src=L-NAT-PUB-2 | | | | |Dest=R-PRIV-2 | | | | |USE-CANDIDATE | | | | | | | | | |(58) Bind Res | | | | |<---------------| | | | |Src=R-PRIV-2 | | | | |Dest=L-NAT-PUB-2| | | | |Map=L-NAT-PUB-2 | | | | | | | | |(59) Bind Res | | | |<--------------------------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-NAT-PUB-2| | | | |Map=L-NAT-PUB-2 | | | | | | | |(60) Bind Res | | | | |<---------------| | | | |Src=R-NAT-PUB-2 | | | | |Dest=L-PRIV-2 | | | | |Map=L-NAT-PUB-2 | | | | | | | | | | | | | | |(61) SIP INVITE | | | | |------------------------------------------------->| | | | | | | | | | |(62) SIP INVITE | | | | |--------------->| | | | | | | | | |(63) SIP 200 OK | | |<-------------------------------------------------| | | | | | |(64) SIP 200 OK | | | | |<---------------| | | | | | | | | |(65) SIP ACK | | | | |------------------------------------------------->| | | | | | | | | | |(66) SIP ACK | | | | |--------------->| | | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 41] Internet-Draft NAT Scenarios September 2008 Figure 13: Endpoint Independent NAT with ICE o On deciding to initiate a SIP voice session the SIP client 'L' starts a local STUN client. The STUN client generates a TURN Allocate request as indicated in (1) from Figure 13 which also highlights the source address and port combination for which the client device wishes to obtain a mapping. The Allocate request is sent through the NAT towards the public internet. o The Allocate message (2) traverses the NAT to the public internet towards the public TURN server. Note that the source address of the Allocate request now represents the public address and port from the public side of the NAT (L-NAT-PUB-1). o The TURN server receives the Allocate request and processes it appropriately. This results in a successful Allocate response being generated and returned (3). The message contains details of the server reflexive address which is to be used by the originating client to receive media (see 'Map=L-NAT-PUB-1') from (3)). It also contains an appropriate TURN-relayed address that can be used at the STUN server (see 'Rel=TURN-PUB-2'). o The Allocate response traverses back through the NAT using the binding created by the initial Allocate request and presents the new mapped address to the client (4). The process is repeated and a second STUN derived set of address' are obtained, as illustrated in (5)-(8) in Figure 13. At this point the User Agent behind the NAT has pairs of derived external server reflexive and relayed representations. The client would be free to gather any number of external representations using any UNSAF[RFC3424] compliant protocol. o The client now constructs a SIP INVITE message (9). The INVITE request will use the addresses it has obtained in the previous STUN/TURN interactions to populate the SDP of the SIP INVITE. This should be carried out in accordance with the semantics defined in the ICE specification[I-D.ietf-mmusic-ice], as shown below in Figure 14: Boulton, Ed., et al. Expires March 21, 2009 [Page 42] Internet-Draft NAT Scenarios September 2008 v=0 o=test 2890844526 2890842807 IN IP4 $L-PRIV-1 c=IN IP4 $L-PRIV-1.address t=0 0 a=ice-pwd:$LPASS a=ice-ufrag:$LUNAME m=audio $L-PRIV-1.port RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=rtcp:$L-PRIV-2.port a=candidate:$L1 1 UDP 2130706431 $L-PRIV-1.address $L-PRIV-1.port typ host a=candidate:$L1 2 UDP 2130706430 $L-PRIV-2.address $L-PRIV-2.port typ host a=candidate:$L2 1 UDP 1694498815 $L-NAT-PUB-1.address $L-NAT-PUB-1.port typ srflx raddr $L-PRIV-1.address rport $L-PRIV-1.port a=candidate:$L2 2 UDP 1694498814 $L-NAT-PUB-2.address $L-NAT-PUB-2.port typ srflx raddr $L-PRIV-1.address rport $L-PRIV-2.port a=candidate:$L3 1 UDP 16777215 $STUN-PUB-2.address $STUN-PUB-2.port typ relay raddr $L-PRIV-1.address rport $L-PRIV-1.port a=candidate:$L3 2 UDP 16777214 $STUN-PUB-3.address $STUN-PUB-3.port typ relay raddr $L-PRIV-1.address rport $L-PRIV-2.port Figure 14: ICE SDP Offer o The SDP has been constructed to include all the available candidates that have been assembled. The first set of candidates (as identified by Foundation $L1) contain two local addresses that have the highest priority. They are also encoded into the connection (c=) and media (m=) lines of the SDP. The second set of candidates, as identified by Foundation $L2, contains the two server reflexive addresses obtained from the STUN server for both RTP and RTCP traffic (identified by candidate-id $L2). This entry has been given a priority lower than the pair $L1 by the client. The third and final set of candidates represents the relayed addresses (as identified by $L3) obtained from the STUN server. This pair has the lowest priority and will be used as a last resort if both $L1 or $L2 fail. o The SIP signaling then traverses the NAT and sets up the SIP session (9)-(10). On advertising a candidate address, the client should have a local STUN server running on each advertised candidate address. This is for the purpose of responding to incoming STUN connectivity checks. o On receiving the SIP INVITE request (10) client 'R' also starts local STUN servers on appropriate address/port combinations and gathers potential candidate addresses to be encoded into the SDP (as the originating client did). Steps (11-18) involve client 'R' carrying out the same steps as client 'L'. This involves Boulton, Ed., et al. Expires March 21, 2009 [Page 43] Internet-Draft NAT Scenarios September 2008 obtaining local, server reflexive and relayed addresses. Client 'R' is now ready to generate an appropriate answer in the SIP 200 OK message (19). The example answer follows in Figure 14: v=0 o=test 3890844516 3890842803 IN IP4 $R-PRIV-1 c=IN IP4 $R-PRIV-1.address t=0 0 a=ice-pwd:$RPASS m=audio $R-PRIV-1.port RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=rtcp:$R-PRIV-2.port a=candidate:$L1 1 UDP 2130706431 $R-PRIV-1.address $R-PRIV-1.port typ host a=candidate:$L1 2 UDP 2130706430 $R-PRIV-2.address $R-PRIV-2.port typ host a=candidate:$L2 1 UDP 1694498815 $R-NAT-PUB-1.address $R-NAT-PUB-1.port typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port a=candidate:$L2 2 UDP 1694498814 $R-NAT-PUB-2.address $R-NAT-PUB-2.port typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port a=candidate:$L3 1 UDP 16777215 $STUN-PUB-2.address $STUN-PUB-4.port typ relay raddr $R-PRIV-1.address rport $R-PRIV-1.port a=candidate:$L3 2 UDP 16777214 $STUN-PUB-3.address $STUN-PUB-5.port typ relay raddr $R-PRIV-1.address rport $R-PRIV-1.port Figure 15: ICE SDP Answer o The two clients have now exchanged SDP using offer/answer and can now continue with the ICE processing - User Agent 'L' assuming the role controlling agent, as specified by ICE. The clients are now required to form their Candidate check lists to determine which will be used for the media streams. In this example User Agent 'L's 'Foundation 1' is paired with User Agent 'R's 'Foundation 1', User Agent 'L's 'Foundation 2' is paired with User Agent 'R's 'Foundation 2', and finally User Agent 'L's 'Foundation 3' is paired with User Agent 'R's 'Foundation 3'. User Agents 'L' and 'R' now have a complete candidate check list. Both clients now use the algorithm provided in ICE to determine candidate pair priorities and sort into a list of decreasing priorities. In this example, both User Agent 'L' and 'R' will have lists that firstly specifies the host address (Foundation $L1), then the server reflexive address (Foundation $L2) and lastly the relayed address (Foundation $L3). All candidate pairs have an associate state as specified in ICE. At this stage, all of the candidate pairs for User Agents 'L' and 'R' are initialized to the 'Frozen' state. The User Agents then scan the list and move the candidates to the 'Waiting' state. At this point both clients will periodically, Boulton, Ed., et al. Expires March 21, 2009 [Page 44] Internet-Draft NAT Scenarios September 2008 starting with the highest candidate pair priority, work their way down the list issuing STUN checks from the local candidate to the remote candidate (of the candidate pair). As a STUN Check is attempted from each local candidate in the list, the candidate pair state transitions to 'In-Progress'. As illustrated in (23), client 'L' constructs a STUN connectivity check in an attempt to validate the remote candidate address received in the SDP of the 200 OK (20) for the highest priority in the check list. As a private address was specified in the active address in the SDP, the STUN connectivity check fails to reach its destination causing a STUN failure. Client 'L' transitions the state for this candidate pair to 'Failed'. In the mean time, Client 'L' is attempting a STUN connectivity check for the second candidate pair in the returned SDP with the second highest priority (24). As can be seen from messages (24) to (29), the STUN Bind request is successful and returns a positive outcome for the connectivity check. Client 'L' is now free to send media to the peer using the candidate pair. Immediately after sending its 200 Okay, Client 'R' also carries out the same set of binding requests. It firstly (in parallel) tries to contact the active address contained in the SDP (30) which results in failure. o In the mean time, a successful response to a STUN connectivity check by User Agent 'R' (27) results in a tentative check in the reverse direction - this is illustrated by messages (31) to (36). Once this check has succeeded, User Agent 'R' can transition the state of the appropriate candidate to 'Succeeded', and media can be sent (RTP). The previously (31-36) described check confirm on both sides (User Agent 'L' and 'R') that connectivity can be achieved using the appropriate candidate pair. User Agent 'L', as the controlling client now sends another connectivity check for the candidate pair, this time including the 'USE-CANDIDATE' attribute as specified in ICE to signal the chosen candidate. This exchange is illustrated in messages (37) to (42). o As part of the process in this example, both 'L' and 'R' will now complete the same connectivity checks for part 2 of the component named for the favored 'Foundation' selected for use with RTCP. The connectivity checks for part '2' of the candidate component are shown in 'L'(43-48) and 'R'(49-54). Once this has succeeded, User Agent 'L' as the controlling client sends another connectivity check for the candidate pair. This time the 'USE- CANDIDATE' attribute is again specified to signal the chosen candidate for component '2'. o The candidates have now been fully verified (and selected) and as they are the highest priority, an updated offer (61-62) is now sent from the offerer (client 'L') to the answerer (client 'R') representing the new active candidates. The new offer would look as follows: Boulton, Ed., et al. Expires March 21, 2009 [Page 45] Internet-Draft NAT Scenarios September 2008 v=0 o=test 2890844526 2890842808 IN IP4 $L-PRIV-1 c=IN IP4 $L-NAT-PUB-1.address t=0 0 a=ice-pwd:$LPASS a=ice-ufrag:$LUNAME m=audio $L-NAT-PUB-1.port RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=rtcp:$L-NAT-PUB-2.port a=candidate:$L2 1 UDP 2203948363 $L-NAT-PUB-1.address $L-NAT-PUB-1.port typ srflx raddr $L-PRIV-1.address rport $L-PRIV-1.port a=candidate:$L2 2 UDP 2172635342 $L-NAT-PUB-2.address $L-NAT-PUB-2.port typ srflx raddr $L-PRIV-1.address rport $L-PRIV-2.port Figure 16: ICE SDP Updated Offer o The resulting answer (63-64) for 'R' would look as follows: v=0 o=test 3890844516 3890842804 IN IP4 $R-PRIV-1 c=IN IP4 $R-PRIV-1.address t=0 0 a=ice-pwd:$RPASS a=ice-ufrag:$RUNAME m=audio $R-PRIV-1.port RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=rtcp:$R-PRIV-2.port a=candidate:$L2 1 UDP 2984756463 $R-NAT-PUB-1.address $R-NAT-PUB-1.port typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port a=candidate:$L2 2 UDP 2605968473 $R-NAT-PUB-2.address $R-NAT-PUB-2.port typ srflx raddr $R-PRIV-1.address rport $R-PRIV-2.port Figure 17: ICE SDP Updated Answer 5.2.2. Address and Port Dependant NAT 5.2.2.1. STUN Failure This section highlights that while using STUN techniques is the preferred mechanism for traversal of NAT, it does not solve every case. The use of basic STUN on its own will not guarantee traversal through every NAT type, hence the recommendation that ICE is the preferred option. Boulton, Ed., et al. Expires March 21, 2009 [Page 46] Internet-Draft NAT Scenarios September 2008 Client PORT/ADDRESS-Dependant STUN [..] NAT Server | | | | |(1) BIND Req | | | |Src=L-PRIV-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | | |(2) BIND Req | | | |Src=NAT-PUB-1 | | | |Dest=STUN-PUB | | | |----------------->| | | | | | | |(3) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=NAT-PUB-1 | | | |Map=NAT-PUB-1 | | | | | | |(4) BIND Resp | | | |<-----------------| | | |Src=STUN-PUB | | | |Dest=L-PRIV-1 | | | |Map=NAT-PUB-1 | | | | | | | |(5)SIP INVITE | | | |------------------------------------------------------->| | | | | | | |(6)SIP 200 OK | | |<------------------------------------| | | | | |(7)SIP 200 OK | | | |<-----------------| | | | | | | |========================================================| |>>>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>| |========================================================| | | | | | x=====================================| | xIncoming Media sent to L-PRIV-1<<<<<<| | x=====================================| | | | | |(8)SIP ACK | | | |----------------->| | | | |(9) SIP ACK | | | |------------------------------------>| | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 47] Internet-Draft NAT Scenarios September 2008 Figure 18: Port/Address-Dependant NAT with STUN - Failure The example in Figure 18 is conveyed in the context of a client behind the 'Port/Address-Dependant' NAT initiating a call. It should be noted that the same problem applies when a client receives a SIP invitation and is behind a Port/Address-Dependant NAT. o In Figure 18 the client behind the NAT obtains a server reflexive representation using standard STUN mechanisms (1)-(4) that have been used in previous examples in this document (e.g Section 5.2.1.1.1). o The external mapped address (server reflexive) obtained is also used in the outgoing SDP contained in the SIP INVITE request(5). o In this example the client is still able to send media to the external client. The problem occurs when the client outside the NAT tries to use the reflexive address supplied in the outgoing INVITE request to traverse media back through the 'Port/Address Dependent' NAT. o A 'Port/Address-Dependant' NAT has differing rules from the 'Endpoint Independent' type of NAT (as defined in RFC4787 [RFC4787]). For any internal IP address and port combination, data sent to a different external destination does not provide the same public mapping at the NAT. In Figure 18 the STUN query produced a valid external mapping for receiving media. This mapping, however, can only be used in the context of the original STUN request that was sent to the STUN server. Any packets that attempt to use the mapped address, that do not originate from the STUN server IP address and optionally port, will be dropped at the NAT. Figure 18 shows the media being dropped at the NAT after (7) and before (8). This then leads to one way audio. 5.2.2.2. TURN Solution As identified in Section Section 5.2.2.1, STUN provides a useful tool for the traversal of the majority of NATs but fails with Port/Address Dependent NAT. The TURN extensions [I-D.ietf-behave-turn] address this scenario. TURN extends STUN to allow a client to request a relayed address at the TURN server rather than a reflexive representation. This then introduces a media relay in the path for NAT traversal (as described in Section 4.2.3.2). The following example explains how TURN solves the previous failure when using STUN to traverse a 'Port/ Address Dependent' type NAT. L Port/Address-Dependant STUN [..] NAT Server | | | | |(1) Alloc Req | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 48] Internet-Draft NAT Scenarios September 2008 |Src=L-PRIV-1 | | | |Dest=STUN-PUB-1 | | | |----------------->| | | | | | | | |(2) Alloc Req | | | |Src=NAT-PUB-1 | | | |Dest=STUN-PUB-1 | | | |----------------->| | | | | | | |(3) Alloc Resp | | | |<-----------------| | | |Src=STUN-PUB-1 | | | |Dest=NAT-PUB-1 | | | |Map=NAT-PUB-1 | | | |Rel=STUN-PUB-2 | | | | | | |(4) Alloc Resp | | | |<-----------------| | | |Src=STUN-PUB-1 | | | |Dest=L-PRIV-1 | | | |Map=NAT-PUB-1 | | | |Rel=STUN-PUB-2 | | | | | | | |(5) Alloc Req | | | |Src=L-PRIV-2 | | | |Dest=STUN-PUB-1 | | | |----------------->| | | | | | | | |(6) Alloc Req | | | |Src=NAT-PUB-2 | | | |Dest=STUN-PUB-1 | | | |----------------->| | | | | | | |(7) Alloc Resp | | | |<-----------------| | | |Src=STUN-PUB-1 | | | |Dest=NAT-PUB-2 | | | |Map=NAT-PUB-2 | | | |Rel=STUN-PUB-3 | | | | | | |(8) Alloc Resp | | | |<-----------------| | | |Src=STUN-PUB-1 | | | |Dest=L-PRIV-2 | | | |Map=NAT-PUB-2 | | | |Rel=STUN-PUB-3 | | | | | | | |(9)SIP INVITE | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 49] Internet-Draft NAT Scenarios September 2008 |----------------->| | | | | | | | |(10)SIP INVITE | | | |------------------------------------>| | | | | | | |(11)SIP 200 OK | | |<------------------------------------| | | | | |(12)SIP 200 OK | | | |<-----------------| | | | | | | |========================================================| |>>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>| |========================================================| | | | | | | |==================| | | |<<| | | |<<<| | | | | | | | |(14) SIP ACK | | | |------------------------------------>| | | | | Figure 19: Port/Address-Dependant NAT with TURN - Success o In this example, client 'L' issues a TURN allocate request(1) to obtained a relay address at the STUN server. The request traverses through the 'Port/Address-Dependant' NAT and reaches the STUN server (2). The STUN server generates an Allocate response (3) that contains both a server reflexive address (Map=NAT-PUB-1) Boulton, Ed., et al. Expires March 21, 2009 [Page 50] Internet-Draft NAT Scenarios September 2008 of the client and also a relayed address (Rel=STUN-PUB-2). The relayed address maps to an address mapping on the STUN server which is bound to the public pin hole that has been opened on the NAT by the Allocate request. This results in any traffic sent to the TURN server relayed address (Rel=STUN-PUB-2) being forwarded to the external representation of the pin hole created by the Allocate request(NAT-PUB-1). o The TURN derived address (STUN-PUB-2) arrives back at the originating client (4) in an Allocate response. This address can then be used in the SDP for the outgoing SIP INVITE request as shown in the following example (note that the example also includes client 'L' obtaining a second relay address for use in the RTCP attribute (5-8)): v=0 o=test 2890844342 2890842164 IN IP4 $L-PRIV-1 c=IN IP4 $STUN-PUB-2.address t=0 0 m=audio $STUN-PUB-2.port RTP/AVP 0 a=rtcp:$STUN-PUB-3.port o On receiving the INVITE request, the UAS is able to stream media and RTCP to the relay address (STUN-PUB-2 and STUN-PUB-3) at the STUN server. As shown in Figure 19 (between messages (12) and (13), the media from the UAS is directed to the relayed address at the STUN server. The STUN server then forwards the traffic to the open pin holes in the Port/Address-Dependant NAT (NAT-PUB-1 and NAT-PUB-2). The media traffic is then able to traverse the 'Port/ Address-Dependant' NAT and arrives back at client 'L'. o TURN on its own will work for 'Port/Address-Dependent' and other types of NAT mentioned in this specification but should only be used as a last resort. The relaying of media through an external entity is not an efficient mechanism for NAT traversal and comes at a high processing cost. 5.2.2.3. ICE Solution The previous two examples have highlighted the problem with using core STUN for all forms of NAT traversal and a solution using TURN for the Address/Port-Dependent NAT case. The RECOMMENDED mechanism for traversing all varieties of NAT is using ICE, as detailed in Section 4.2.3.3. ICE makes use of core STUN, TURN and any other UNSAF[RFC3424] compliant protocol to provide a list of prioritized addresses that can be used for media traffic. Detailed examples of ICE can be found in Section 5.2.1.2.1. These examples are associated with an 'Endpoint-Independent' type NAT but can be applied to any NAT type variation, including 'Address/Port-Dependant' type NAT. The ICE Boulton, Ed., et al. Expires March 21, 2009 [Page 51] Internet-Draft NAT Scenarios September 2008 procedures carried out are the same. For a list of candidate addresses, a client will choose where to send media dependant on the results of the STUN connectivity checks and associated priority (highest priority wins). It should be noted that the inclusion of a NAT displaying Address/Port-Dependent properties does not automatically result in relayed media. In fact, ICE processing will avoid use of media relay with the exception of two clients which both happen to be behind a NAT using Address/Port-Dependent characteristics. The connectivity checks and associated selection algorithm enable traversal in this case. Figure 20 and following description provide a guide as to how this is achieved using the ICE connectivity checks. This is an abbreviated example that assumes successful SIP offer/answer exchange and illustrates the connectivity check flow. L Port/Address-Dependent Endpoint-Independent R L-NAT R-NAT |========================================================| | SIP OFFER/ANSWER EXCHANGE | |========================================================| | | | | | | |(1)Bind Req | | | |<-----------------| | | |Src=R=PRIV-1 | | | |Dest=L-NAT-PUB-1 | | | | | | |(2)Bind Req | | | x<-----------------| | | |Src=R-NAT-PUB-1 | | | |Dest=L-NAT-PUB-1 | | | | | | |(3)Bind Req | | | |----------------->| | | |Src=L-PRIV-1 | | | |Dest=R-NAT-PUB-1 | | | | | | | | |(4)Bind Req | | | |----------------->| | | |Src=L-NAT-PUB-1 | | | |Dest=R-NAT-PUB-1 | | | | | | | | |(5)Bind Req | | | |----------------->| | | |Src=L-NAT-PUB-1 | | | |Dest=R-PRIV-1 | | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 52] Internet-Draft NAT Scenarios September 2008 | | |(6)Bind Resp | | | |<-----------------| | | |Src=R-PRIV-1 | | | |Dest=L-NAT-PUB-1 | | | | | | |(7)Bind Resp | | | |<-----------------| | | |Src=R-NAT-PUB-1 | | | |Dest=L-NAT-PUB-1 | | | | | | |(8)Bind Resp | | | |<-----------------| | | |Src=R-NAT-PUB-1 | | | |Dest=L-PRIV-1 | | | | | | | | | |(9)Bind Req | | | |<-----------------| | | |Src=R-Priv-1 | | | |Dest=L-NAT-PUB-1 | | |(10)Bind Req | | | |<-----------------| | | |Src=R-NAT-PUB-1 | | | |Dest=L-NAT-PUB-1 | | | | | | |(11)Bind Req | | | |<-----------------| | | |Src=R-NAT-PUB-1 | | | |Dest=L-PRIV-1 | | | | | | | |(12)Bind Resp | | | |----------------->| | | |Src=L-PRIV-1 | | | |Dest=L-NAT-PUB-1 | | | | | | | | |(13)Bind Resp | | | |----------------->| | | |Src=L-NAT-PUB-1 | | | |Dest=R-NAT-PUB-1 | | | | | | | | |(14)Bind Resp | | | |----------------->| | | |Src=L-NAT-PUB-1 | | | |Dest=R-PRIV-1 | | | | | | Boulton, Ed., et al. Expires March 21, 2009 [Page 53] Internet-Draft NAT Scenarios September 2008 Figure 20: Single Port/Address-Dependant NAT - Success In this abbreviated example, Client R has already received a SIP INVITE request and is starting its connectivity checks with Client L. Client R generates a connectivity check (1) and sends to client L's information as presented in the SDP offer. The request arrives at client L's Port/Address dependent NAT and fails to traverse as there is no NAT binding. This would then move the connectivity check to a failed state. In the mean time client L has received the SDP answer in the SIP request and will also commence connectivity checks. A check is dispatched (3) to Client R. The check is able to traverse the NAT due to the association set up in the previously failed check(1). The full Bind request/response is shown in steps (3)-(8). As part of a candidate pair, Client R will now successfully be able to complete the checks, as illustrated in steps (9)-(14). The result is a successful pair of candidates that can be used without the need to relay any media. In conclusion, the only time media needs to be relayed is a result of clients both behind Address/Port Dependant NAT type. As you can see from the example in this section, neither side would be able to complete connectivity checks with the exception of the Relayed candidates. 6. IPv4-IPv6 Transition This section describes how IPv6-only SIP user agents can communicate with IPv4-only SIP user agents. While the techniques discussed in this draft primarily contain examples of traversing NATs to allow communications between hosts in private and public networks, they are by no means limited to such scenarios. The same NAT traversal techniques can also be used to establish communication in a heterogeneous network environment -- e.g., communication between an IPv4 host and an IPv6 host. 6.1. IPv4-IPv6 Transition for SIP Signaling IPv4-IPv6 translations at the SIP level usually take place at dual- stack proxies that have both IPv4 and IPv6 DNS entries. Since this translations do not involve NATs that are placed in the middle of two SIP entities, they fall outside the scope of this document. A detailed description of this type of translation can be found in [I-D.ietf-sipping-v6-transition] Boulton, Ed., et al. Expires March 21, 2009 [Page 54] Internet-Draft NAT Scenarios September 2008 6.2. IPv4-IPv6 Transition for Media Figure 21 shows a network of IPv6 SIP user agents that has a relay with a pool of public IPv4 addresses. The IPv6 SIP user agents of this IPv6 network need to communicate with users on the IPv4 Internet. To do so, the IPv6 SIP user agents use TURN to obtain a set of public IPv4 address and port pairs from the relay (for RTP and RTCP). The mechanism that an IPv6 SIP user agent follows to obtain public IPv4 address and port pairs from a relay using TURN is the same as the one followed by a user agent with a private IPv4 address to obtain public IPv4 address and port pairs. The example below explains how a UA in an IPv6-only network can use ICE [I-D.ietf-mmusic-ice] to communicate with a SIP Phone in an IPv4-only network. Note that no server reflective addresses are used in this example. +----------+ | / \ | /SIP \ /Phone \ / \ ------------ | | | | 192.0.2.2:25000 | | 192.0.2.2:25123 RTP RTCP +-------------+ | TURN Server | +-------------+ IPv4 Network | | +---------+ | | ----------------------| NAT |--------------------- | | +---------+ IPv6 Network | | | | | | [2001:DB8::1]:30000 RTP RTCP [2001:DB8::1]:30001 +----------+ | IPv6 SIP | | UA | +----------+ Boulton, Ed., et al. Expires March 21, 2009 [Page 55] Internet-Draft NAT Scenarios September 2008 Figure 21: IPv6-IPv4 transition scenario The IPv6 UA obtains a TURN-derived IPv4 address and port pair for its RTP port and another one for its RTCP port by issuing 2 TURN Allocate requests. The TURN server generates responses containing relayed IPv4 addressee and port pairs for both RTP and RTCP ports. These IPv4 addresses and port pairs map to the IPv6 source addressee and port pairs. The result of any UDP packets sent to the IPv4 address and port pairs provided by the TURN server (i.e., 192.0.2.2:25000 for RTP and 192.0.2.2:25123 for RTCP) with be redirected to the IPv6 IP address and port pairs of the SIP UA (i.e., [2001:DB8::1]:30000 for RTP and [2001:DB8::1]:30001 for RTCP). When the UA builds the original Offer, it includes 2 candidates: one for the host IPv6 address and another for the relay IPv4 address. When computing the priority for the candidate, we will use a type preference of 126 for the host address candidate, and of 0 for the relay address candidate, a local preference of 65535 for both candidates, and a component ID of 1 for RTP and 2 for RTCP for both candidates. This will generate a priority of 2130706431 for the host address, and of 16777215 for the relay address. The default candidate is the relay address candidate. The Offer will look as follows. v=0 o=test 2890844342 2890842164 IN IP6 2001:DB8::1 c=IN IP4 192.0.2.2 t=0 0 a=ice-pwd:asd88fgpdd777uzjYhagZg a=ice-ufrag:8hhY m=audio 25000 RTP/AVP 0 a=rtcp:25123 a=candidate:1 1 UDP 2130706431 [2001:DB8::1] 30000 typ host a=candidate:1 2 UDP 2130706430 [2001:DB8::1] 30001 typ host a=candidate:2 1 UDP 16777215 192.0.2.2 25000 typ relay raddr [2001:DB8::1] rport 30000 a=candidate:2 2 UDP 16777214 192.0.2.2 25123 typ relay raddr [2001:DB8::1] rport 30001 The Offer is sent in an INVITE request which gets routed to the IPv4- only UA, which will choose the IPv4 candidate as per normal ICE procedures. 7. Security Considerations There are no Security Considerations beyond the ones inherited by Boulton, Ed., et al. Expires March 21, 2009 [Page 56] Internet-Draft NAT Scenarios September 2008 reference. 8. IANA Considerations There are no IANA Considerations. 9. IAB Considerations There are no IAB considerations. 10. Acknowledgments The authors would like to thank the members of the IETF SIPPING WG for their comments and suggestions. Expert review and contribution was provided by Francois Audet. Detailed comments were provided by Vijay Gurbani, kaiduan xie, Remi Denis-Courmont, Hadriel Kaplan, Phillip Matthews, Dan Wing, Spencer Dawkins and Hans Persson. 11. References 11.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3261] 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. [RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol (SIP): Locating SIP Servers", RFC 3263, June 2002. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC2766] Tsirtsis, G. and P. Srisuresh, "Network Address Boulton, Ed., et al. Expires March 21, 2009 [Page 57] Internet-Draft NAT Scenarios September 2008 Translation - Protocol Translation (NAT-PT)", RFC 2766, February 2000. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002. [RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing", RFC 3581, August 2003. [RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts", RFC 3327, December 2002. [RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne, "Grouping of Media Lines in the Session Description Protocol (SDP)", RFC 3388, December 2002. [RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)", RFC 3605, October 2003. [RFC4787] Audet, F. and C. Jennings, "Network Address Translation (NAT) Behavioral Requirements for Unicast UDP", BCP 127, RFC 4787, January 2007. [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, July 2007. [I-D.ietf-sip-connect-reuse] Mahy, R., Gurbani, V., and B. Tate, "Connection Reuse in the Session Initiation Protocol (SIP)", draft-ietf-sip-connect-reuse-11 (work in progress), July 2008. [I-D.ietf-behave-rfc3489bis] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for (NAT) (STUN)", draft-ietf-behave-rfc3489bis-18 (work in progress), July 2008. [I-D.ietf-behave-turn] Rosenberg, J., Mahy, R., and P. Matthews, "Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)", draft-ietf-behave-turn-09 (work in progress), July 2008. Boulton, Ed., et al. Expires March 21, 2009 [Page 58] Internet-Draft NAT Scenarios September 2008 [I-D.ietf-sip-outbound] Jennings, C. and R. Mahy, "Managing Client Initiated Connections in the Session Initiation Protocol (SIP)", draft-ietf-sip-outbound-15 (work in progress), June 2008. [I-D.ietf-sip-gruu] Rosenberg, J., "Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)", draft-ietf-sip-gruu-15 (work in progress), October 2007. [I-D.ietf-mmusic-ice] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", draft-ietf-mmusic-ice-19 (work in progress), October 2007. [I-D.ietf-avt-rtp-and-rtcp-mux] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress), August 2007. 11.2. Informative References [I-D.ietf-sipping-sbc-funcs] Hautakorpi, J., Camarillo, G., Penfield, B., Hawrylyshen, A., and M. Bhatia, "Requirements from SIP (Session Initiation Protocol) Session Border Control Deployments", draft-ietf-sipping-sbc-funcs-06 (work in progress), June 2008. [I-D.ietf-mmusic-media-path-middleboxes] Stucker, B. and H. Tschofenig, "Analysis of Middlebox Interactions for Signaling Protocol Communication along the Media Path", draft-ietf-mmusic-media-path-middleboxes-01 (work in progress), July 2008. [I-D.ietf-sipping-v6-transition] Camarillo, G., "IPv6 Transition in the Session Initiation Protocol (SIP)", draft-ietf-sipping-v6-transition-07 (work in progress), August 2007. [RFC3424] Daigle, L. and IAB, "IAB Considerations for UNilateral Self-Address Fixing (UNSAF) Across Network Address Translation", RFC 3424, November 2002. Boulton, Ed., et al. Expires March 21, 2009 [Page 59] Internet-Draft NAT Scenarios September 2008 Authors' Addresses Chris Boulton Avaya Eastern Business Park St Mellons Cardiff, South Wales CF3 5EA Email: cboulton@avaya.com Jonathan Rosenberg Cisco Systems 600 Lanidex Plaza Parsippany, NJ 07054 Email: jdrosen@cisco.com Gonzalo Camarillo Ericsson Hirsalantie 11 Jorvas 02420 Finland Email: Gonzalo.Camarillo@ericsson.com Francois Audet Nortel 4655 Great America Parkway Santa Clara CA 95054 US Email: audet@nortel.com Boulton, Ed., et al. Expires March 21, 2009 [Page 60] Internet-Draft NAT Scenarios September 2008 Full Copyright Statement Copyright (C) The IETF Trust (2008). 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. 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, THE IETF TRUST 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. Intellectual Property 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. Boulton, Ed., et al. Expires March 21, 2009 [Page 61]