Integration of Resource Management and Call Signaling for IP Telephony
draft-dcsgroup-mmusic-resource-00

Versions: 00                                                            
MMUSIC Working Group                                        W. Marshall
Internet Draft                                          K. Ramakrishnan
Document: <draft-dcsgroup-mmusic-resource-00.txt>                  AT&T
Category: Informational
                                                              E. Miller
                                                             G. Russell
                                                              CableLabs

                                                               B. Beser
                                                            M. Mannette
                                                        K. Steinbrenner
                                                                   3Com

                                                                D. Oran
                                                                  Cisco

                                                             J. Pickens
                                                                  Com21

                                                            P. Lalwaney
                                                             J. Fellows
                                                     General Instrument

                                                               D. Evans
                                                           Lucent Cable

                                                               K. Kelly
                                                               NetSpeak

                                                           F. Andreasen
                                                              Telcordia

                                                             June, 1999


 Integration of Resource Management and Call Signaling for IP Telephony


Status of this Memo

   This document is an Internet-Draft and is NOT offered in accordance
   with Section 10 of RFC2026 [1], and the author does not provide the
   IETF with any rights other than to publish as an Internet-Draft.

   Internet-Drafts are working documents of the Internet Engineering
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   documents at any time. It is inappropriate to use Internet- Drafts
   as reference material or to cite them other than as "work in
   progress."



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   The list of current Internet-Drafts can be accessed at
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   The distribution of this memo is unlimited.  It is filed as <draft-
   dcsgroup-mmusic-resource-00.txt>, and expires December 31, 1999.
   Please send comments to the authors.



1. Abstract

   This document describes a two-stage call-setup mechanism, with the
   resource management protocol interleaved between the two stages. The
   objective of such a mechanism is to enable deployment of robust IP
   Telephony services. The first phase ensures that resources are made
   available before the phone rings and the participants of the call
   are "invited" to participate. The second phase involves the alerting
   of the user after the network resources are reserved. This draft
   presents the call signaling requirements that lead to this design,
   and the specific SIP extension of an INVITE that does not ring the
   far-end phone.


2. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in
   this document are to be interpreted as described in RFC-2119 [2].


3.  Introduction

   For an Internet Telephony service to be successfully used by a large
   number of subscribers, it must offer few surprises to those
   accustomed to the behavior of existing telephony services.  One
   expectation is that of connection quality, implying resources must
   be set aside for each call.

   A key contribution of the DCS architecture, as described in [3], is
   a recognition of the need for coordination between call signaling,
   which controls access to telephony specific services, and resource
   management, which controls access to network-layer resources. This
   coordination is designed to meet the user expectations and human
   factors associated with telephony.

   While customers may expect, during times of heavy load, to receive a
   "fast busy" or an announcement saying "all circuits are busy now,"
   the general expectation is that once the destination phone rings
   that the connection can be made.  Blocking a call after ringing the



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   destination is considered a "call defect" and is a very undesirable
   exception condition.

   We consider the case where a provider may choose to block a call
   when adequate resources for the call are not available. It may be
   argued that best-effort connections may be an alternative in such a
   case. However, public policy demands that the phone system provide
   adequate quality at least in certain cases: e.g., for emergency
   communications during times of disasters.  Call blocking enables a
   provider to meet such requirements. This draft and the overall DCS
   architecture assumes that the provider blocks calls when resources
   are unavailable.

   It is often the case that calls into a disaster area are blocked, to
   ensure resources are available for calls out of the disaster area.
   Such policy-level controls also need to be available for the service
   provider.

   A further expectation of customers of a carrier-grade telephony
   system is that the first few syllables of a conversation not be
   clipped.  While residential users often introduce a long delay
   between pickup and use of the voice path (e.g., time taken to lift
   receiver and placing it to the ear before speaking), automatic
   answering devices and operators with headsets have no such delay and
   demand fast cut-through of the voice path.

   Coordination between call signaling and resource management is also
   needed to prevent fraud and theft of service.  The coordination
   between call-signaling and QoS setup protocols ensures that users
   are authenticated and authorized before receiving access to the
   enhanced QoS associated with the telephony service.

3.1 Requirements

   The basic motivation in this work is to meet and possibly exceed the
   user expectations and human factors associated with telephony.

   In this section, we briefly describe the application requirements
   that led to the set of DCS signaling design principles.  In its
   basic implementation, DCS supports a residential telephone service
   comparable to the local telephone services offered today. Some of
   the requirements for resource management, in concert with call
   signaling, are as follows:

   The system must minimize call defects.  These are situations where
   either the call never completes, or an error occurs after the
   destination is alerted.  Requirements on call defects are typically
   far more stringent than call blocking.  Note that we expect the
   provider and the endpoints to attempt to use lower bandwidth CODECs
   as the first line of defense against limited network capacity, and
   to avoid blocking calls.




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   The system must minimize the post-dial-delay, which is the time
   between the user dialing the last digit and receiving positive
   confirmation from the network.  This delay must be short enough that
   users do not perceive a difference with post-dial delay in the
   circuit switched network or conclude that the network connectivity
   no longer exists.

   The system must minimize the post-pickup-delay, which is the time
   between the user picking up a ringing phone and the voice path being
   ready for use.  This must be short enough that the "hello" is not
   clipped (both from the called party and the caller).  In practical
   terms, this delay cannot be more than tens of milliseconds beyond
   the one-way propagation delay.

   Call signaling needs to provide enough information to the resource
   management protocol so as to enable resources to be allocated in the
   network.  This typically requires most if not all of the components
   of a packet classifier (source IP, destination IP, source port,
   destination port, protocol) to be available for resource allocation.


3.2 Overview

   For acceptable interactive voice communication it is important to
   achieve end-to-end QoS. The end-to-end QoS assurance implies
   achieving low packet delay and packet loss. End-to-end packet delay
   must be small enough that it does not interfere with normal voice
   conversations. The ITU recommends no greater than 300 ms roundtrip
   delay for telephony service.  Packet loss must be small enough to
   not perceptibly impede voice quality or the performance of fax and
   voice band modems.

   If it is found that the network does not have end-to-end QoS
   resources there are two alternatives: either (1) allow call
   signaling to proceed with high probability of excessive delay and
   packet loss which could impair any interactive real-time
   communication between the participants, or (2) to block the call
   prior to the called party being alerted.  When calls are blocked
   because of a lack of resources in a particular segment of the
   network, it is highly desirable that such blocking occur before the
   called party picks up.

   We do expecte the endpoints to attempt to use lower bandwidth
   CODECs, thereby avoiding blocking calls, as the first line of
   defense against limited network capacity.

   The call-signaling and resource reservation must be achieved in such
   a way that the post-dial-delay must be minimized without increasing
   the probability of call defects. This means that the number of
   round-trip messages must be kept at an absolute minimum and messages
   must be send directly end system-to-end system if possible.




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   Since the post-pickup-delay has a much more stringent requirement
   than the post-dial-delay, post-pickup-delay must not depend on any
   signals that are not sent directly end-to-end. Otherwise it may be
   difficult to avoid quality impairments such as clipping of the
   initial user "hello".


3.3 Basic Call Flow

   Originating (MTA-o)                          Terminating (MTA-t)
        |               SIP-Proxy(s)                    |
        |  INVITE(NO-RING)      |                       |
        |---------------------->|---------------------->|
        |                       |                       |
        |                       |       200 OK          |
        |<----------------------|<----------------------|
        |                                               |
        |                                               |
        |                       ACK                     |
        |---------------------------------------------->|
        | Reservation                       Reservation |
         ===========>                       <===========
        |                                               |
        |                                               |
        |               INVITE(RING)                    |
        |---------------------------------------------->|
        |
        |
        |                                       User Alerted
        |                       RINGING                 |
        |<----------------------------------------------|
        |                                               |
        |                                       User Picks-Up
        |                                       the phone
        |                       200 OK                  |
        |<----------------------------------------------|
        |
        |
        |                       ACK                     |
        |---------------------------------------------->|

                                Figure 1


   Figure 1 presents a high-level overview of a basic end-point to end-
   point(MTA-to-MTA) call flow.

   When a user goes off-hook and dials a telephone number, the
   originating MTA (MTA-o) collects the dialed digits and sends the
   initial call INVITE message to a SIP-Proxy.

   Upon reception the initial INVITE message at the destination MTA
   (MTA-t), the MTA invokes call feature handling, such as call-


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   forwarding but does not alert the user.  This action is indicated by
   the existence of the NO-RING statement.

   Assuming that the call is not forwarded, MTA-t communicates the
   coding style and bandwidth requirements for the media streams.  The
   200 OK response to the initial INVITE is forwarded back through the
   SIP-proxies.

   The INVITE request and 200 OK response contain a SIP Contact header
   to indicate the IP address of the remote MTA to be used for
   subsequent end-to-end SIP signaling exchanges.  MTA-o acknowledges
   the 200 OK directly to MTA-t.

   Once the initial INVITE/200 OK exchange has completed, each MTA has
   sufficient information regarding the other end-point, bandwidth and
   other characteristics of the media exchange. Now the endpoints may
   reserve the resources that will be needed for the media streams.

   Once MTA-o has successfully made its reservation, it sends a second
   INVITE message directly to MTA-t without the NO-RING statement,
   stating that the MTA-t should alert the user.  If MTA-t successfully
   reserved the resources needed for the call, it responds with a 180
   Ringing to indicate that the user is alerted, and that the calling
   user should be given a ring-back call progress tone.  When the
   called party answers, by going off-hook, MTA-t sends a 200 OK final
   response directly to MTA-o, which MTA-o acknowledges.

   Either party can terminate the call.  An MTA that detects an "on-
   hook" (request to terminate the call) releases the QoS resources
   held for the connection, and sends a SIP BYE message to the remote
   MTA, which is acknowledged.


4.  Changes to SIP to Support Two-Stage Call Setup

   The profile/extension defined in this section allows network
   resources to be reserved prior to alerting the user and
   simultaneously reduces the user perceived delays.

   The following syntax specification uses the augmented Backus-Naur
   Form (BNF) as described in RFC-2234 [4].


4.1 SIP Header Extension: NO-RING

   The NO-RING request header allows the caller to indicate that an
   INVITE should not alert the user.

        NO-RING         = "NoRing:"

   A SIP server, on receiving an INVITE with "NO-RING" MUST execute the
   feature logic associated with the "line" (the logical end-point) and
   seize the line if it is not busy and the number has not been


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   forwarded, but the server SHOULD NOT alert the user.  Once network
   resources have been reserved, the originating client MUST issue a
   second INVITE without the NO-RING header. This second INVITE
   instructs the destination server to alert the user.


4.2 SIP Profile

   This section defines a SIP [RFC 2543] profile for integrating
   resource management.

4.2.1 Phase One

   Phase one consists of an INVITE message which does not alert the
   user.  The INVITE message can be sent through proxies or (in the
   case that the desired end point is known, and no resource
   reservation needs to be authorized by the network, and enhanced QoS
   is not desired for communication) to the destination itself. Because
   of the possibility of call forwarding, the originator does not know
   the final end-point and its characteristics (e.g. CODECs supported,
   etc.); therefore the resource reservation MUST NOT be attempted yet.

   Additionally, to support CODEC selection an SDP session description
   MUST be included in:
   @ The INVITE message MUST contain the NO_RING header.

   @ The 200 OK response to the INVITE MUST contain the NO_RING header,
     and a CONTACT header that indicates the final recipient of the
     call.

   @ The ACK associated with INVITE containing the NO-RING header MUST
     be send end-to-end.


4.2.2
      Phase Two

   The message sent during phase two depends on the result of the QoS
   reservation carried out after phase one. If the reservation is
   unsuccessful the originating MTA sends a BYE message to terminate
   the session.

   If the reservation is successful, the originating MTA sends a second
   INVITE message directly to the destination MTA, to the address as
   indicated by the CONTACT header. This direct end-to-end INVITE
   message is necessary to minimize post-pickup delay. The 200 OK final
   response message generated after the called user picks up will also
   be sent directly end-to-end to minimize post-pickup delay as well as
   avoid clipping the initial "hello". Otherwise, there is the
   potential of excessive delay if the 200 OK goes through proxies
   after the phone is picked up, resulting in a delay to cut-through
   the voice path.




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   The second INVITE and 200 OK messages MUST NOT contain the NO-RING
   statement since during this phase the user should be alerted. The
   INVITE message MUST NOT carry different SDP descriptions than that
   communicated in the first phase.

   Upon reception of the INVITE message, the terminating MTA responds
   based upon its QoS reservation. If the reservation is unsuccessful
   then it MUST return an error indication.  If the reservation is
   successful, the terminating MTA MUST return 180 ringing message and
   alert the user, or return a 200 OK message directly when the called
   user picks up.

5. Advantages of the Proposed Approach

   The use of two-phase call signaling makes it possible for SIP to
   meet user expectations that come from the telephony services.

   The reservation of resources before the user is alerted makes sure
   that the network resources are assured before the destination end-
   point is informed about the call.

   The number of messages and the total delay introduced by these
   messages is kept to a minimum without sacrificing compatibility with
   classic SIP.

   Because the response messages to second phase goes directly end-to-
   end between the endpoints without traversing the DCS-proxies, the
   post-dial-delay is minimized. This reduces the chance that the
   "hello" greeting of the caller (in response to the callee's
   greeting) gets clipped.


6. Security Considerations

   There are no security implication recognized by the extensions
   proposed in this draft.


7. References


   1  Bradner, S., "The Internet Standards Process -- Revision 3", BCP
      9, RFC 2026, October 1996.

   2  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997

   3  DCS Group, "Architectural Considerations for Providing Carrier
      Class Telephony Services Utilizing SIP-based Distributed Call
      Control Mechanisms", draft-dcsgroup-mmusic-arch-00.txt, June
      1999.




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   4  Crocker, D. and Overell, P.(Editors), "Augmented BNF for Syntax
      Specifications: ABNF", RFC 2234, Internet Mail Consortium and
      Demon Internet Ltd., November 1997






8.
  Acknowledgments

   The Distributed Call Signaling work in the PacketCable project is
   the work of a large number of people, representing many different
   companies.  The authors would like to recognize and thank the
   following for their assistance: John Wheeler, Motorola; David
   Boardman, Daniel Paul, Arris Interactive; Bill Blum, Jon Fellows,
   Jay Strater, Jeff Ollis, Clive Holborow, General Instruments; Doug
   Newlin, Guido Schuster, Ikhlaq Sidhu, 3Com; Jiri Matousek, Bay
   Networks; Farzi Khazai, Nortel; John Chapman, Bill Guckle, Cisco;
   and Chuck Kalmanek, Doug Nortz, John Lawser, James Cheng, and Partho
   Mishra, AT&T.


9. Author's Addresses

   Bill Marshall
   AT&T
   Florham Park, NJ  07932
   Email: wtm@research.att.com

   K. K. Ramakrishnan
   AT&T
   Florham Park, NJ  07932
   Email: kkrama@research.att.com

   Ed Miller
   CableLabs
   Louisville, CO  80027
   Email: E.Miller@Cablelabs.com

   Glenn Russell
   CableLabs
   Louisville, CO  80027
   Email: G.Russell@Cablelabs.com

   Burcak Beser
   3Com
   Rolling Meadows, IL  60008
   Email: Burcak_Beser@3com.com

   Mike Mannette
   3Com

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   Rolling Meadows, IL  60008
   Email: Michael_Mannette@3com.com

   Kurt Steinbrenner
   3Com
   Rolling Meadows, IL  60008
   Email: Kurt_Steinbrenner@3com.com

   Dave Oran
   Cisco
   Acton, MA  01720
   Email: oran@cisco.com

   John Pickens
   Com21
   San Jose, CA
   Email: jpickens@com21.com

   Poornima Lalwaney
   General Instrument
   San Diego, CA  92121
   Email: plalwaney@gi.com

   Jon Fellows
   General Instrument
   San Diego, CA  92121
   Email: jfellows@gi.com

   Doc Evans
   Lucent Cable Communications
   Westminster, CO  30120
   Email: n7dr@lucent.com

   Keith Kelly
   NetSpeak
   Boca Raton, FL  33587
   Email: keith@netspeak.com

   Flemming Andreasen
   Telcordia
   Piscataway, NJ
   Email: fandreas@telcordia.com












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