Signaling End-Of-Stream in RTSP (or RTCP)

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Signaling End-Of-Stream in RTSP(or RTCP)
IETF-58 MMUSIC WG
In IETF I-D database draft-ietf-zeng-rtsp-end-of-
stream-00.txt (on mmusic mailing list
draft-ietf-zeng-mmusic-end-of-stream-01.txt)

• Thomas Zeng
• P. Greg Sherwood
• PacketVideo Corp, Nov, 2003

2
Background

• Work triggered by the awkwardness of BYE
• Tasked to create Strawman proposal in Vienna
• Lately this work has been energized by the
  desire for server to push different types of
  information to client
• END_OF_STREAM with ending packet number
• Session descriptions (and their updates)
• Error events
• Another background Sean Sheedy had a announce
  based EOS draft a few years ago, and Ron
  Frederick had mentioned Stream done event in
  the RTSP bake-off report as well (about 3 years
  ago)
• http//www.ietf.org/proceedings/00jul/00july-131.h
  tm

3
Background (contd)

• BYE RTCP packet has been widely used for
  streaming server to announce EOS event
• EOS the end of stream is reached and/or play
  request has been fulfilled.
• Drawbacks of using BYE
• BYE de-allocates SSRC, thus preventing re-play
  w/o SETUP again
• BYE packet may be unreliable (e.g., when
  carried over UDP)
• BYE cannot not be aggregated
• If BYE is not used and server does nothing to
  signal EOS, then
• Clients rely on the close Range to figure out
  when RTP packets are over
• If no end range, which is the case for live
  streaming (or end range is provided in PLAY
  response but server has to end early due to
  error)
• The clients transport handler will know if no
  RTP packet is forth-coming, but the cost can be
  one or more of the following
• extra delay and/or erroneous requests for
  re-transmission
• even re-buffering if video has low frame rate

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Strawman Proposal Using RTSP

• As an extension to RTSP, use a S-gtC method w/ a
  feature tag
• Recent suggestion is to use ANNOUNCE instead of a
  new method name
• This method announces EOS event, with optional
  information on actual range served and the
  sequence number(s) of the ending RTP packet(s)
• MUST include Notice header ( a new header) for
  event type
• Do we need SDP as body of this method?
• Request URL can be either aggregate or
  non-aggregate URI
• Opportunity exists to generalize this as a method
  for server to push other kind of info to client
• Need to work out how server can initiate
  connection to client

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Example of RTSP conversation

• S-gtC ANNOUNCE rtsp//foo.com/bar.avi/streamid0
  RTSP/1.0
• CSeq 10
• Session 12345678
• Notice End-Of-Stream
• Reason 2000 End of range reached
• Range npt0-200 bytes0-200000
• RTP-Info url//foo.com/bar.avi/streamid
  0seq45102
• C-gtS RTSP/1.0 200 OK
• CSeq 10
• Session 12345678
• / Second play can reuse SSRC /
• C-gtS PLAY rtsp//foo.com/bar.avi/streamid0
  RTSP/1.0
• Supported method.eos
• CSeq 110
• Session 12345678
• Range 0-200

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Strawman Proposal Using RTCP

• Define a new RTCP packet type (type 205), which
  has format similar to BYE (but added ending seq
  no. per SSRC), except it does not mean leaving
  the RTP session
• Instead, it means I am going to stop sending RTP
  packet until you issue a new PLAY
• RTCP extension really belongs to AVT WG !
• 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
  1 2 3 4 5 6 7 8 9 0 1
• ----------------------
  ----------
• V2P SC PTEOS205
  length
• ----------------------
  ---------- ----
• SSRC/CSRC
  repeat
• ----------------------
  ---------- SC times
• extended RTP sequence number
  
• ----------------------
  ---------- ----
• ...
  
• 
  
• (opt) length reason for
  EOS ...
• ----------------------
  ----------

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Pros and Cons of The Two Proposals
RTSP END_OF_STREAM RTCP EOS Packet
Works for non-persistent RTSP connection ? No, unless server can initiate connection (TBD) Yes
Aggregatable? Yes No
Reliable? Yes, assuming RTSP over TCP No, unless RTP/AVP/TCP is used
Works for non-RTP transport (e.g., Mpeg2 transport) Yes No
Scalable in multicast transports? No Yes
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What Next

• Propose to expand the scope of this draft to
  cover ANNOUNCE as an extension to RTSP core spec
• Pushes events such as EOS
• Pushes Session Descriptions as did RFC2326
• Change name of draft to rtsp-announce-
• Work out how server contact client host
• Dual-roled host client to upstream, server to
  downstream
• Acceptable as MMUSIC WG work item?
• ANNOUNCE has been taken out of RTSP core spec
  (rfc2326bis)