Subject: Protocol Action: 'Aggregation of RSVP Reservations over MPLS TE/DS-TE Tunnels' to Proposed Standard

From: The IESG <[email-masked]>
Date: Tue, 14 Nov 2006 10:52:29 -0500

The IESG has approved the following document:

- 'Aggregation of RSVP Reservations over MPLS TE/DS-TE Tunnels '
   <draft-ietf-tsvwg-rsvp-dste-05.txt> as a Proposed Standard

This document is the product of the Transport Area Working 
Group. 

The IESG contact persons are Magnus Westerlund and Lars Eggert.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-rsvp-dste-05.txt

Technical Summary

This document defines how one can aggregate RSVP reservations when
entering traffic engineered (TE) MPLS tunnels. The MPLS tunnel head-end
act as an aggregator and tunnels the end-to-end RSVP reservation to the
tail-end that is the deaggregator of the reservation. The aggregator is
responsible to ensure that the RSVP reservation is fulfilled through the
tunnel. The aggregator have the knowledge about the current commitment and
behavior of the MPLS TE tunnel, and are thus able to grant or deny further
requests for resource from the aggregate, the MPLS TE tunnel represent.
This mechanism provides benefits from both RSVP aggregation and MPLS
traffic engineering.

Working Group Summary

There is strong consensus in the WG to publish this document. It has been
reviewed by several people including expert reviewers in the WG last call.
Comments raised has been addressed.

Protocol Quality

This document has been well reviewed in the WG and comments raised has
been addressed. PROTO shepherd is James Polk. Responsible AD was Magnus
Westerlund.

Note to RFC Editor
 
Appendix B, Page 28:

DELETE the following text:

   Our example environment relies of [SIP-RSVP] to synchronize RSVP
   bandwidth reservations with SIP. For example, the RSVP bandwidth
   requests may be integrated into the call setup flow as follows (See
   call setup flow diagram in Figure A2):

      - Caller C1 initiates a call by sending a SIP INVITE to VoIP
        gateway GW1, which passes the INVITE along to the call control
        agent (CCA).  The INVITE message may contain a list of codecs
        that the calling phone can support.

      - VoIP gateway GW2, chooses a compatible codec from the list and
        responds with a SIP message 183 Session Progress.

      - When GW1 receives the SIP response message with the SDP, it
        determines how much bandwidth is required for the call.

      - GW1 sends an RSVP Path message to PE1, requesting bandwidth for
        the call.

      - GW2 also sends an RSVP Path message to PE2.

      - Assuming that the tunnel (from left to right) has sufficient
        bandwidth, PE1 responds to GW1 with a Resv message

      - Again assuming the tunnel (from right to left) has sufficient
        bandwidth, PE2 responds to GW2 with a Resv message

      - GW2 sends a SIP 200 OK message to GW1.

      - GW1 sends a SIP UPDATE message to GW2.

      - Upon receiving the UPDATE, GW2 sends the INVITE to the
        destination phone, which responds with SIP message 180 RINGING.

      - When (and if) the called party answers, the destination phone
        responds with another SIP 200 OK which completes the connection
        and tells the calling party that there is now reserved
        bandwidth in both directions so that conversation can begin.

Le Faucheur, et al.                                          [Page 28]

                RSVP Aggregation over MPLS TE tunnels  September 2006

      - RTP media streams in both directions pass through the DSTE
        tunnels as they traverse the MPLS network.

Le Faucheur, et al.                                          [Page 29]

                RSVP Aggregation over MPLS TE tunnels  September 2006

    IP-Phone/                                                  IP-Phone/
     TA-C1      GW1     PE1         CCA          PE2      GW2      TA-C2
     |     INVITE|(SDP1) |  INVITE   |   INVITE   |        |           |
     |---------->|-------|---------->|------------|------->|           |
     |        100|TRYING |           |            |        |           |
     |<----------|-------|-----------|            |        |           |
     |        183|(SDP2) |           |            |        |           |
     |<----------|-------|-----------|------------|--------|           |
     |           | PATH  |           |            |  PATH  |           |
     |           |------>|           |            |<-------|           |
     |           | RESV  |           |            |  RESV  |           |
     |           |<------|           |            |------->|           |
     |           |       |     UPDATE|(SDP3)      |        |           |
     |           |-------|-----------|------------|------->|           |
     |           |       |     200 OK|(SDP4)      |        |           |
     |           |<------|-----------|------------|--------|  INVITE   |
     |           |       |           |            |        |---------->|
     |180 RINGING|       |        180|RINGING     |        |180 RINGING|
     |<----------|<------|-----------|------------|--------|<----------|
     | 200 OK    |    200|OK         |         200|OK      |  200 OK   |
     |<----------|<------|-----------|<-----------|--------|<----------|
     |           |       |           |            |        |           |
     |           |       |       DSTE|TUNNEL      |        |           |
     |        RTP|MEDIA  |-----------|------------|        |           |
     |===========|=======|===========|============|========|==========>|
     |           |       |-----------|------------|        |           |
     |           |       |           |            |        |           |
     |           |       |-----------|------------|        |           |
     |<==========|=======|===========|============|========|===========|
     |           |       |-----------|------------|        |           |
                                 DSTE TUNNEL

           Figure A2. VoIP QoS CAC using SIP with Preconditions

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