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INTERNET-DRAFT                                   13 July 2000
                                                Colin Perkins
                                                      USC/ISI

                 RTP Interoperability Statement

               draft-ietf-avt-rtp-interop-03.txt



                     Status of this memo



This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.

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.

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The list of current Internet-Drafts can be accessed at
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Distribution of this document is unlimited.

Comments are solicited and should be addressed to the author and/or the
IETF Audio/Video Transport working group's mailing list at rem-conf@es.net.



                           Abstract


     It is required to demonstrate interoperability of RTP implementations
     in order to move the RTP specification to draft standard.
     This memo outlines those features to be tested, as the first
     stage of an interoperability statement.

1  Introduction


The Internet standards process [1] places a number of requirements
on a standards track protocol specification.  In particular, when
advancing a protocol from proposed standard to draft standard it
is necessary to demonstrate at least two independent and interoperable
implementations, from different code bases, of all options and features
of that protocol.  Further, in cases where one or more options or
features have not been demonstrated in at least two interoperable
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implementations, the specification may advance to the draft standard
level only if those options or features are removed.  The Real-time
Transport Protocol, RTP, was originally specified in RFC1889 as a
proposed standard [2].  The revision of this specification for draft
standard status is now well underway, so it has become necessary
to conduct such an interoperability demonstration.

This memo describes the set of features and options of the RTP
specification which need to be tested as a basis for this demonstration.
Due to the nature of RTP there are necessarily two types of test described:
those which directly affect the interoperability of implementations at a
``bits on the wire level'' and those which affect scalability and safety of
the protocol but do not directly affect interoperability.  A related memo
[4] describes a testing framework which may aid with interoperability
testing.

This memo is for information only and does not specify a standard
of any kind.

2  Features and options required to demonstrate interoperability


In order to demonstrate interoperability it is required to produce
a statement of interoperability for each feature noted below.  Such
a statement should note the pair of implementations tested, including
version numbers, and a pass/fail statement for each feature.  It
is not expected that every implementation will implement every feature,
but each feature needs to be demonstrated by some pair of applications.

Note that some of these tests depend on the particular profile used,
or upon options in that profile.  For example, it will be necessary
to test audio and video applications operating under [3] separately.



  1. Interoperable exchange of data packets using the basic RTP header
     with no header extension, padding or CSRC list.

  2. Interoperable exchange of data packets which use padding.

  3. Interoperable exchange of data packets which use a header extension.
     There are three possibilities here:  a) if both implementations
     use a header extension in the same manner, it should be verified
     that the receiver correctly receives the information contained
     in the extension header; b) If the sender uses a header extension
     and the receiver does not, it should be verified that the receiver
     ignores the extension; c) If neither implementation implements
     an extended header, this test is considered a failure.

  4. Interoperable exchange of data packets using the marker bit as
     specified in the profile.



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  5. Interoperable exchange of data packets using the payload type
     field to differentiate multiple payload formats according to
     a profile definition.

  6. Interoperable exchange of data packets containing a CSRC list.

  7. Interoperable exchange of RTCP packets, which must be compound
     packets containing at least an initial SR or RR packet and an
     SDES CNAME packet.  Other RTCP packet types may be included,
     but this is not required for this test.

  8. Interoperable exchange of sender report packets when the receiver
     of the sender reports is not also a sender (ie:  sender reports
     which only contain sender info, with no report blocks).

  9. Interoperable exchange of sender report packets when the receiver
     of the sender reports is also a sender (ie:  sender reports
     which contain one or more report blocks).

 10. Interoperable exchange of receiver report packets.

 11. Interoperable exchange of receiver report packets when not receiving
     data (ie:  the empty receiver report which has to be sent first
     in each compound RTCP packet when no-participants are transmitting
     data).

 12. Interoperable and correct choice of CNAME, according to the rules
     in the RTP specification and profile (applications using the
     audio/video profile [3] under IPv4 should typically generate
     a CNAME of the form `example@10.0.0.1', or `10.0.0.1' if they
     are on a machine which no concept of usernames).

 13. Interoperable exchange of source description packets containing
     a CNAME item.

 14. Interoperable exchange of source description packets containing
     a NAME item.

 15. Interoperable exchange of source description packets containing
     an EMAIL item.

 16. Interoperable exchange of source description packets containing
     a PHONE item.

 17. Interoperable exchange of source description packets containing
     a LOC item.

 18. Interoperable exchange of source description packets containing
     a TOOL item.

 19. Interoperable exchange of source description packets containing
     a NOTE item.

 20. Interoperable exchange of source description packets containing
     a PRIV item.



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INTERNET-DRAFT                                   13 July 2000
 21. Interoperable exchange of BYE packets containing a single SSRC.

 22. Interoperable exchange of BYE packets containing multiple SSRCs.

 23. Interoperable exchange of BYE packets containing the optional
     reason for leaving text.

 24. Interoperable exchange of BYE packets containing the optional
     reason for leaving text and multiple SSRCs.

 25. Interoperable exchange of application defined RTCP packets.  As
     with the RTP header extension this test takes two forms:  if
     both implementations implement the same application defined packet
     it should be verified that those packets can be interoperably
     exchanged.  If only one implementation uses application defined
     packets, it should be verified that the other implementation
     can receive compound RTCP packets containing an APP packet whilst
     ignoring the APP packet.  If neither implementation implements
     APP packets this test is considered a failure.

 26. Interoperable exchange of encrypted RTP packets using DES encryption
     in CBC mode.

 27. Interoperable exchange of encrypted RTCP packets using DES encryption
     in CBC mode.

3  Features and options relating to scalability


In addition to the basic interoperability tests, RTP includes a number
of features relating to scaling of the protocol to large groups.
Since these features are those which have undergone the greatest
change in the update of the RTP specification, it is considered important
to demonstrate their correct implementation.  However, since these
changes do not affect the bits-on-the-wire behaviour of the protocol,
it is not possible to perform a traditional interoperability test.
As an alternative to such testing we require that multiple independent
implementations complete the following demonstrations.



  1. Demonstrate correct implementation of basic RTCP transmission
     rules:  periodic transmission of RTCP packets at the minimum
     (5 second) interval and randomisation of the transmission interval.

  2. Demonstrate correct implementation of the RTCP step join backoff
     algorithm as a receiver.

  3. Demonstrate correct implementation of the RTCP step join backoff
     algorithm as a sender.

  4. Demonstrate correct steady state scaling of the RTCP interval
     acording to the group size.

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INTERNET-DRAFT                                   13 July 2000
  5. Demonstrate correct steady state scaling of the RTCP interval
     acording to the group size with compensation for the number of
     senders.

  6. Demonstrate correct implementation of the RTCP reverse reconsideration
     algorithm.

  7. Demonstrate correct implementation of the RTCP BYE reconsideration
     algorithm.

  8. Demonstrate correct implementation of the RTCP member timeout
     algorithm.

  9. Demonstrate random choice of SSRC.

 10. Demonstrate random selection of initial RTP sequence number.

 11. Demonstrate random selection of initial RTP timestamp.

 12. Demonstrate correct implementation of the SSRC collision/loop
     detection algorithm.

 13. Demonstrate correct generation of reception report statistics
     in SR/RR packets.

 14. Demonstrate correct generation of the sender info block in SR
     packets.

4  Author's Address

Colin Perkins
USC Information Sciences Institute
4350 North Fairfax Drive
Suite 620
Arlington, VA 22203
USA


Email:  csp@isi.edu



5  Acknowledgments


Thanks to Steve Casner, Jonathan Rosenberg and Bill Fenner for their
helpful feedback.

6  References



[1] S. Bradner, ``The Internet Standards Process -- Revision 3'',
RFC2026, Internet Engineering Task Force, October 1996.

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[2] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, ``RTP:
A Transport Protocol to Real-Time Applications'', RFC1889, Internet
Engineering Task Force, January 1996.

[3] H. Schulzrinne, ``RTP Profile for Audio and Video Conferences
with Minimal Control'', draft-ietf-avt-profile-new-08.txt, January
2000.

[4] C. S. Perkins, J. Rosenberg and H. Schulzrinne, ``RTP Testing
Strategies'', draft-ietf-avt-rtptest-03.txt, July 2000.


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