Internet DRAFT - draft-ranarang-sipping-middialog-sip-status

draft-ranarang-sipping-middialog-sip-status







Network Working Group                                          R. Narang
Internet-Draft                                        Cisco Systems Inc.
Intended status: Standards Track                               June 2006
Expires: December 3, 2006


  Session Initiation Protocol (SIP) Mid-Dialog Status code for network
                   disconnection on one side of B2BUA
           draft-ranarang-sipping-middialog-sip-status-00.txt

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   This Internet-Draft will expire on December 3, 2006.

Copyright Notice

   Copyright (C) The Internet Society (2006).













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Abstract

   This document defines a new SIP status code needed for communicating
   about the network failure happening in signaling connection on one
   side of the B2BUA to the other side of B2BUA on an established
   dialog/session.  The network failure condition MAY be detected by
   session timers [rfc4028] or any other proprietary mechanism.


Table of Contents

   1.  Requirements notation  . . . . . . . . . . . . . . . . . . . .  3
   2.  Introduction and Overview  . . . . . . . . . . . . . . . . . .  4
     2.1.  Simple Handling of the n/w failure . . . . . . . . . . . .  4
     2.2.  Optimized Handling . . . . . . . . . . . . . . . . . . . .  5
   3.  B2BUA detecting the network failure behavior . . . . . . . . .  6
     3.1.  B2BUA wants to remain in partial dialog for CDR  . . . . .  6
       3.1.1.  UA4 is a non-refresher . . . . . . . . . . . . . . . .  6
       3.1.2.  UA4 is a refresher . . . . . . . . . . . . . . . . . .  6
     3.2.  B2BUA doesn't want to remain in partial dialog . . . . . .  6
   4.  B2BUA receiving the 450 status code behavior . . . . . . . . .  7
     4.1.  UA3 handling . . . . . . . . . . . . . . . . . . . . . . .  7
     4.2.  UA2 handling . . . . . . . . . . . . . . . . . . . . . . .  7
   5.  Phone detecting the network failure behavior . . . . . . . . .  8
     5.1.  Phone2 is a Media Application  . . . . . . . . . . . . . .  8
   6.  Phone receiving the 450 status code behavior . . . . . . . . .  9
     6.1.  450 receipt in session refresh . . . . . . . . . . . . . .  9
     6.2.  450 receipt in BYE's Reason header . . . . . . . . . . . .  9
   7.  Proxy's 450 status code behavior . . . . . . . . . . . . . . . 10
     7.1.  Call Stateful Proxy  . . . . . . . . . . . . . . . . . . . 10
     7.2.  Transaction Stateful Proxy . . . . . . . . . . . . . . . . 10
     7.3.  Stateless Proxy  . . . . . . . . . . . . . . . . . . . . . 10
   8.  Call Stateful Proxy Session Timeout handling . . . . . . . . . 11
   9.  IP-PSTN GW receiving the 450 status code behavior  . . . . . . 12
   10. NAT/ALG's 450 status code behavior . . . . . . . . . . . . . . 13
   11. CDR logging of 450 status code . . . . . . . . . . . . . . . . 14
   12. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
     12.1. IANA Registration of the 450 (Call in minimal state)
           Response Code  . . . . . . . . . . . . . . . . . . . . . . 15
   13. Backward's compatibilty  . . . . . . . . . . . . . . . . . . . 16
   14. 450 (Call in minimal state) Definition . . . . . . . . . . . . 17
   15. Security Considerations  . . . . . . . . . . . . . . . . . . . 18
   16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 20
   Intellectual Property and Copyright Statements . . . . . . . . . . 21






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1.  Requirements notation

   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].














































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2.  Introduction and Overview

   Consider that a SIP session was established using an INVITE dialog
   from phone1 to phone2 with B2BUA's in the path as per the following
   topology:-


       +----+          +---------+        +---------+        +----+
       | +-+|          |+-+   +-+|        |+-+   +-+|        |+-+ |
       | | ||  dialog1 || |   | || dialog2|| |   | || dialog3|| | |
       | |U||<-------->||U|   |U||<------>||U|   |U||<------>||U| |
       | |A||  nw1     ||A|   |A||  nw2   ||A|   |A||   nw3  ||A| |
       | |1||          ||2|   |3||        ||4|   |5||        ||6| |
       | | ||          || |   | ||        || |   | ||        || | |
       | +-+|          |+-+   +-+|        |+-+   +-+|        |+-+ |
       +----+          +---------+        +---------+        +----+
       phone1             B2BUA1            B2BUA2           phone2

                                 Figure 1

   There will be an active INVITE dialogs between individual UA's.
   Consider during session establishment, UA's have negotiated the
   session timers on the individual INVITE dialogs as per [rfc4028].
   The session timers are actively running for monitoring the liveliness
   of the SIP session.

   Note:- In the B2BUA configuration as above, the session timers are
   independently negotiated for each individual dialog between UA's.
   There is only one media session i.e from phone1 to phone2.

   The session i.e media path from phone1 to phone2 could have been
   established using a different network than the call signaling (i.e
   dialog) path.

   Consider 'nw3' as per above topology encounters a sudden network
   failure condition.  After session timeout for dialog3, the 'UA5' &
   'UA6' will be able to detect this error condition.

2.1.  Simple Handling of the n/w failure

   A simple implementation of UA5 or UA6 would assume this as a critical
   condition.  The UA6 may stop the session (media) and give reorder
   tone to the user.  The UA5 would assume that sip session is also torn
   down, so it will initiate tearing down the dialog2 towards B2BUA1 by
   sending a BYE.






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2.2.  Optimized Handling

   An optimized implementation at UA6 (phone2) would be to not assume
   this as a critical problem for an active SIP media session between
   phone1 & phone2.  It could try its best to 'not' interrupt the active
   media connection between phone2 and phone1.  The UA6 MAY want to wait
   for actual physical hang-up from the phone2 user.

   An optimized implementation at B2BUA2 would be to not tear down an
   active sip 'dialog2' towards B2BUA1 and instead communicate the
   status of nw3's "out of order condition" which impacted dialog3.

   The B2BUA1 on learning about dialog3 disconnection due to n/w
   failure, could take an appropriate action whether to tear down the
   'associated' SIP session towards UA1 or wait for normal hang-up from
   phone1.

   As it is described the requirement is to communicate the failure
   status across the B2BUA about one SIP dialog 'dialog3' onto the
   another SIP dialog 'dialog2' associated with the same SIP session.

   Currently in SIP specification there is no appropriate Status code
   for communicating this kind of failure from one sip dialog to another
   for an established session.

   To fulfill this requirement a new SIP Status code 450 "Call in
   minimal state" is being proposed.

   Note:- The above optimized handling is also possible when there is
   only one B2BUA in the configuration.





















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3.  B2BUA detecting the network failure behavior

   This is referring UA5 in Figure 1.  The UA5 shall be monitoring the
   session by using session timers or some other proprietary mechanism.
   We shall be discussing in the context of session timers throughout
   this document.  During session establishment, it would have been
   negotiated whether UA5 is a "refresher" or a "non-refresher".

   The failure of session timer's refresh would not certainly tell
   whether UA6 crashed or it is due to network connectivity issues.  For
   optimized handling UA5 SHALL make an assumption that session timeout
   is due to network connectivity issues.  The UA5 SHALL internally
   communicate this condition to the UA4 by passing the 450 status code.
   Then UA5 MAY terminate itself.

   The UA4 MAY handle this special condition in two ways depending on
   whether or not it needs to remain in the partial dialog for the
   purposes of logging "Call Detail Records" on completion of the SIP
   session with phone1.

3.1.  B2BUA wants to remain in partial dialog for CDR

3.1.1.  UA4 is a non-refresher

   When UA4 is the "non-refresher" for dialog2, then on receiving the
   next session refresh i.e a re-INVITE or UPDATE, a 450 "Call in
   minimal state" response MUST be sent.

3.1.2.  UA4 is a refresher

   When UA4 is the "refresher" for dialog2, then an INVITE/UPDATE with
   Reason header having SIP Status code 450 MUST be sent.

   Reason: SIP;cause=450;text="Call in minimal state"

   The session refresh procedure on dialog2 MUST be continued until a
   BYE is received from remote or session refresh timeout occurs.

3.2.  B2BUA doesn't want to remain in partial dialog

   A BYE with SIP Status code 450 SHOULD be sent and UA4 MUST be
   terminated on receiving 200 OK (for BYE).

   Reason: SIP;cause=450;text="Call in minimal state"







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4.  B2BUA receiving the 450 status code behavior

   This is referring to UA3 in Figure 1.  The receipt of SIP Status
   code="450" by UA3 would indicate that somewhere in the network
   segment connecting from remote UA or further down, the network is
   detected as "disconnected".  There is no specific information whether
   the SIP Session/media path between the caller and callee is still
   active or got disconnected.  The received 450 status code MUST be
   communicated to the other side of B2BUA.

4.1.  UA3 handling

   The UA3 MUST interwork the 450 status code to the UA2 in the same
   B2BUA.  The UA3 handling SHALL be based on whether BYE with 450 in
   Reason header is received or whether it is the refresher and 450
   response to re-INVITE/UPDATE is received or whether it is the non-
   refresher and re-INVITE/UPDATE with 450 in Reason header is received.

   The receipt of BYE SHOULD immediately terminate the UA3 and dialog2.

   Otherwise the session timers mechanism on dialog2 MUST continue until
   UA2 sends a disconnect (for receipt of BYE on dialog1, or when UA2
   needs to tear down the partial dialog by sending a BYE on dialog1).
   If session timer's timeout occurs on UA3 then also UA3 MUST clear
   itself.

4.2.  UA2 handling

   The UA2 handling is going to be same as UA4 as it is described above.
   The UA2 MAY handle this special condition in two ways depending on
   whether or not it needs to remain in the partial dialog for the
   purposes of logging "Call Detail Records" on completion of the SIP
   session with phone1.

   Preferably immediate B2BUA interacting with the phone UA SHOULD NOT
   clear the dialog with phone UA until hangup from the phone.  The
   session refresh SHOULD continue until then.

   If UA2 needs to clear itself, it MUST send an internal disconnect to
   UA3 and MUST send a BYE with Reason header having a SIP Status
   code=450 on dialog1.  After UA2 receives 200 OK (for BYE) it MUST
   terminate itself.  The internal disconnect to UA3 SHALL result in
   sending BYE on dialog2, which SHALL clear the dialog with remote UA4.

   If UA2 needs to remain then it SHOULD continue with the session
   timers procedure and respond with 450 status code for non-refresher
   case and MUST send INVITE/UPDATE having 450 in the reason header for
   refresher case on the session timeout.



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5.  Phone detecting the network failure behavior

   This is referring UA6 (phone2) in Figure 1.  When a phone detects a
   network failure condition for an established signaling connection via
   session timers or any other proprietary mechanism it implies a
   problem in the connectivity for the SIP dialog.  The associated SIP
   media session could have been established through a different
   network, so sip dialog disconnection due to n/w error doesn't
   definitely imply that media session is also impacted.

   Rather than tearing down the media session, best attempt handling
   would be to just preserve the existing session in a limited mode i.e
   no other feature capabilities possible.  For disconnecting the
   session, the phone MAY wait for hang-up from the user or MAY depend
   on media specific monitoring of the session.

   In this condition the SIP dialog SHALL be terminated immediately, but
   SIP media session MAY be preserved until hang-up.  So phone2 SHALL go
   into a state where it will just have the media session operational.

   There SHALL be no 450 status code generation by the phone UA as SIP
   dialog is already torn down.

5.1.  Phone2 is a Media Application

   It is possible that phone2 is a Media Application i.e. there is no
   human user e.g IVR or Call Recorder.  On session timeout at UA6
   (phone2), the SIP dialog with immediate B2BUA SHALL be cleared.  So
   there SHALL be no BYE message received to tear down the media
   session.

   If Media application is implementing media session preservation for
   such scenarios, it MUST be required to additionally kick-off RTP/RTCP
   monitoring capability on such a condition, so that it is detected
   when to stop communicating media with the remote and close the RTP/
   RTCP channels.















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6.  Phone receiving the 450 status code behavior

   This is referring UA1 (phone1) in Figure 1.  The receipt of 450
   status code SHALL intimate to the phone that the network through
   which the sip dialog was established has detected an error in
   connectivity.  There is no definite information about the status of
   SIP media session.

   An optimized action at the UA1 would be to do the best effort
   handling i.e no interruption to the existing media session.  For
   making sure no features on the SIP dialog are allowed, the softkeys
   on the phone MUST be disabled.

6.1.  450 receipt in session refresh

   If UA1 is the refresher, the 450 response SHALL be received in
   response to INVITE or UPDATE.  If UA1 is the non-refresher, the 450
   response SHALL be received in the INVITE's Reason header.  The
   session timer's refresh procedure MUST keep on running.

   For tearing down the session the phone UA MUST wait for hang-up from
   the user.  When user hangs up then a BYE message MUST be sent to the
   B2BUA.

6.2.  450 receipt in BYE's Reason header

   This is the scenario when B2BUA's doesn't want to remain in the
   minimal dialog for CDR purposes.  The Phone UA MUST interpret the
   Reason header in BYE and if 450 is received it would mean to attempt
   optimized handling as described above.  The phone UA MUST send 200 OK
   for BYE and dialog1 SHALL be terminated.

   The phone1 will go into a state where it will just have the media
   session operational.  On user hangup the media session i.e RTP/RTCP
   channels SHOULD be closed.
















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7.  Proxy's 450 status code behavior

   Consider there is a Proxy in the signaling path from phone1 to
   phone2.

7.1.  Call Stateful Proxy

   Consider that a sip dialog was established from UA through the proxy
   to a remote UA, the proxy chose to be in the path of all the requests
   sent in a sip dialog by adding Record-Route header in the requests
   and responses forwarded through it.

   For the mid dialog re-INVITE's forwarded through the Proxy, if the
   client transaction receives a 450 status code from the UAS, the same
   450 response MUST be forwarded by the proxy to the server transaction
   for sending to the UAC.  The standard handling as per section 16.7 of
   per [rfc3261] SHALL be performed with no special handling specific to
   450.

7.2.  Transaction Stateful Proxy

   The proxy which is only a transaction stateful will not be in the
   path of SIP messages after initial INVITE/200 OK exchange.  The 450
   status code is intended to be useful on an established session/dialog
   for communicating about sudden network failure in the signaling path
   in some segment.  The 450 status code SHALL be generated by B2BUA and
   possibly will not be traversing through the proxy, which is
   transaction stateful only.

7.3.  Stateless Proxy

   If 450 status code response is received by a stateless Proxy, the
   standard handling as per section 16.11 of [rfc3261] SHALL be
   performed.

















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8.  Call Stateful Proxy Session Timeout handling

   Assume the configuration having UA endpoints communicating through a
   call stateful proxy and no B2BUA in the path.  During session
   establishment the session timers were exchanged and actively running
   for monitoring the session.

   There will be a session timer running on the stateful proxy.  If this
   timeout occurs on the proxy it would mean either the refresher UA
   hasn't refreshed or 200 OK from non-refresher isn't received in time.
   It might be possible that n/w path from proxy to either UA got lost.
   On encountering this condition, the proxy will remove the call state
   w/o initiating the BYE from itself as per [rfc4028].  In case later
   an INVITE/UPDATE/BYE or other response is received from the UA's, it
   shall be proxied to the other side without using the call stateful
   proxy logic i.e will be routed by using the Request-URI in the SIP
   message.

   The proxy SHALL NOT generate a 450 status code on its own in this
   condition.

   At the UA endpoints also, there will be a session timeout condition
   i.e refresher will not receive 200 OK or non-refresher will not
   receive the session refresh request.  The handling for this condition
   at the phone UA will be implementation specific or as per section 5
   of this draft.  As both UA endpoints will autonomously detect this
   error condition, either side SHALL NOT be generating or handling 450
   status code in this situation.























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9.  IP-PSTN GW receiving the 450 status code behavior

   The receipt of 450 status code SHALL imply that remote UA has
   probably detected a network disconnection in a sip dialog towards
   other side of the UA or further down the network towards the
   destination party, a disconnection caused sending of 450 status code
   on the sip dialog towards the IP-PSTN GW.

   The 450 status code could be received in the following ways:-

   1.  BYE with Reason header having 450 status code.  With receipt of
   BYE the dialog from IP-GW to the remote UA gets terminated.

   2.  SIP leg on the GW is the refresher and in response to re-INVITE/
   UPDATE, a 450 status code is received.  The session timers MAY
   continue until hang-up from the user on the PSTN side or until a
   failure occurs in session refresh procedure.

   3.  SIP leg on the GW is the non-refresher and a remote sends a re-
   INVITE with Reason header having 450 status code.  The session timers
   MAY continue until hang-up from the user on the PSTN side or until a
   failure occurs in session refresh procedure.

   For the above cases the IP-PSTN GW MAY maintain a SIP leg/dialog in a
   minimal state allowing the preservation of the established media
   session.  There MAY NOT be any notification to the PSTN side about
   this situation.  Any features attempted from the PSTN side MUST be
   rejected with appropriate Q.931 specific error codes.























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10.  NAT/ALG's 450 status code behavior

   If NAT receives a 450 status code during session refresh, it would
   imply that network link in the further segment has been detected
   down.  The session refresh between the sip elements will be
   continuing until a BYE is received, so there SHOULD NOT be transport
   disconnection on receipt of 450 by NAT/ALG's.












































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11.  CDR logging of 450 status code

   If there has been 450 status code specific handling in a SIP call leg
   then while CDR's are generated the information about 450 MAY be
   logged to CDR's.  This will indicate that media might have been
   active longer than the call duration logged in the CDR's due to the
   specific nature of the 450 status code, allowing media session to be
   active without the SIP dialog for n/w error conditions.











































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12.  IANA Considerations

   This extension defines a new response code i.e 450.

12.1.  IANA Registration of the 450 (Call in minimal state) Response
       Code

   The following is the registration for the 450 (Call in minimal state)
   response code:

   Response Code: 450

   Default Reason Phrase: Call in minimal state

   RFC Number:




































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13.  Backward's compatibilty

   If a version of intermediate proxy or B2BUA is not implementing 450
   response handling then it will result in session timeout on that
   element, which will result in immediate clearing of the established
   dialog/media session for the call.













































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14.  450 (Call in minimal state) Definition

   This response code indicates that the network through which the sip
   dialog was successfully established has detected an error in
   connectivity.  There is no definite information about the status of
   SIP session.  Its default reason phrase is "Call in minimal state".













































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15.  Security Considerations

   None.
















































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16.  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", June 2002.

   [rfc4028]  Donovan, S. and J. Rosenberg, "Session Timers in the
              Session Initiation Protocol (SIP)", April 2005.








































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Author's Address

   Rajeev Narang
   Cisco Systems Inc.
   12 Bannerghatta Road
   Subramanya Arcade Block D
   Bangalore
   India

   Email: ranarang@cisco.com









































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