Network Working Group | M. Boucadair |
Internet-Draft | France Telecom |
Intended status: Informational | April 16, 2014 |
Expires: October 18, 2014 |
PCP for SIP Deployments in Managed Networks
draft-boucadair-pcp-sip-ipv6-01
This document discusses how PCP (Port Control Protocol) can be used in SIP deployments in managed networks.
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The base PCP specification allows to retrieve the external IP address and external port to be conveyed in the SIP signaling messages [RFC3261]. Therefore SIP Proxy Servers do not need to support means to ease the NAT traversal of SIP messages (e.g., [RFC5626], [RFC6223], etc.). Another advantage of using the external IP address and port is this provides a hint to the proxy server there is no need to return a small expire timer (e.g., 60s). In addition, the outbound proxy does not need any further feature to be supported in order to assist the remote endpoint to successfully establish media sessions. In particular, ALGs are not required in the NAT for this purpose and no dedicated functions at the media gateway are needed.
This document discusses how PCP can be used in SIP deployments (including IPv6 considerations).
The benefits of using PCP for SIP deployments are listed below:
Experimentation results, including SIP flow examples, are documented in [I-D.boucadair-pcp-nat64-experiments].
Even in deployments where ICE [RFC5245] is required, PCP can be of great help as discussed in [I-D.penno-rtcweb-pcp].
The document is targeting SIP deployments in managed networks.
The PCP base specification allows to create mappings in PCP-controlled devices and therefore prepare for receiving incoming packets. A SIP User Agent can use PCP to create one mapping for SIP signalling messages and other mappings for media session purposes.
The SIP UA uses the external IP address and port number to build SIP headers. In particular, this information is used to build the VIA and CONTACT headers.
The external IP address and port(s) instantiated for media streams, are used to build the SDP offer/answer. In particular, the "c" line and "m" lines.
PCP allows to discover and to set the lifetime of mapping instantiated in intermediate middleboxes.
As a consequence of instantiating mappings for media/session flows, incoming packets can be successfully forwarded to the appropriate SIP UA. Particularly, unidirectional media flows (e.g., announcement server) will be forwarded accordingly.
For deployments relying on classic RTP/RTCP odd/even port numbers assignment scheme, PORT_SET option [I-D.ietf-pcp-port-set] can be use by a SIP UA to request port parity be preserved by the PCP server.
For deployments assuming RTCP port number can be deduced from the RTP port number, PORT_SET option [I-D.ietf-pcp-port-set] can be used by a SIP UA to retrieve a pair of contiguous ports from the PCP server.
If the SIP UA is located behind a NAT64 device [RFC6146], the option defined in [I-D.ietf-pcp-nat64-prefix64] can be used to retrieve the PREFIX64 used by that NAT64 device. The retrieved prefix will be used to locally build an IPv6-converted IPv4 address corresponding to the IPv4 address included in the SDP message received from a remote IPv4-only SIP UA in a SDP offer or answer.
The base PCP specification can be used to retrieve the port number to be singled if "a=rtcp" attribute is in use [RFC3550].
PCP can be used to discover the DSCP value to be used when sending real-time flows or to create a mapping that matches a DSCP marking. This can be achieved using the DSCP option defined in [I-D.boucadair-pcp-extensions]. DSCP setting value is configured by the network and not the SIP UA.
This feature can be used as an input for DSCP marking in some deployments such as [I-D.ietf-tsvwg-rtcweb-qos].
Because an IPv6-only SIP UA is not aware of the connectivity capabilities of the remote UA, the IPv6-only SIP UA uses the ALTC attribute to signal the assigned IPv6 address and the IPv4 address learned via PCP. If the remote SIP UA is IPv6-enabled, IPv6 transfer capabilities will be used to place the session. If the remote SIP UA is IPv4-only, IPv4 transfer capabilities will be used. NAT64 devices will be crossed only if the remote UA is IPv4-only.
SIP UAs co-located with the B4 [RFC6333] or located behind the CPE can behave as dual-stack UAs:
To avoid unnecessary invocation of AFTR resources, ALTC attribute is used to signal both IPv4 and IPv6 addresses. If the remote SIP UA is IPv6-enabled, IPv6 transfer capabilities will be used to place the session (i.e., the flows will avoid crossing the DS-Lite AFTR device). If the remote SIP UA is IPv4-only, IPv4 transfer capabilities will be used. AFTR devices will be crossed only if the remote UA is IPv4-only.
PCP-related security considerations are discussed in [RFC6887].
This document does not require any action from IANA.
Many thanks for T. Reddy and S. Kiesel for their review.
[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. |
[RFC3550] | Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. |
[RFC3581] | Rosenberg, J. and H. Schulzrinne, "An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing", RFC 3581, August 2003. |
[RFC6887] | Wing, D., Cheshire, S., Boucadair, M., Penno, R. and P. Selkirk, "Port Control Protocol (PCP)", RFC 6887, April 2013. |