rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt[49]

[入库:2006年2月23日] [更新:2007年3月24日]

本文简介:


            THEN (optionally) count occurrence of own traffic looped.
                 mark current time in conflicting address list entry.
                 ABORT processing of data packet or control element.
       log occurrence of a collision.
       create a new entry in the conflicting address list and
       mark current time.
       send an RTCP BYE packet with the old SSRC identifier.
       choose a new identifier.
       create a new entry in the source identifier table with the
         old SSRC plus the source transport address from the packet
         being processed.
       CONTINUE with normal processing.

   In this algorithm, packets from a newly conflicting source address
   will be ignored and packets from the original source will be kept.
   (If the original source was through a mixer and later the same source
   is received directly, the receiver may be well advised to switch
   unless other sources in the mix would be lost.) If no packets arrive
   from the original source for an extended period, the table entry will
   be timed out and the new source will be able to take over. This might
   occur if the original source detects the collision and moves to a new
   source identifier, but in the usual case an RTCP BYE packet will be
   received from the original source to delete the state without having
   to wait for a timeout.

   When a new SSRC identifier is chosen due to a collision, the
   candidate identifier should first be looked up in the source
   identifier table to see if it was already in use by some other
   source. If so, another candidate should be generated and the process
   repeated.

   A loop of data packets to a multicast destination can cause severe
   network flooding. All mixers and translators are required to
   implement a loop detection algorithm like the one here so that they
   can break loops. This should limit the excess traffic to no more than
   one duplicate copy of the original traffic, which may allow the
   session to continue so that the cause of the loop can be found and
   fixed. However, in extreme cases where a mixer or translator does not
   properly break the loop and high traffic levels result, it may be
   necessary for end systems to cease transmitting data or control
   packets entirely. This decision may depend upon the application. An
   error condition should be indicated as appropriate. Transmission
   might be attempted again periodically after a long, random time (on
   the order of minutes).

Schulzrinne, et al          Standards Track                    [Page 48]

RFC 1889                          RTP                       January 1996


9.  Security

   Lower layer protocols may eventually provide all the security
   services that may be desired for applications of RTP, including
   authentication, integrity, and confidentiality. These services  have
   recently been specified for IP. Since the need for a confidentiality
   service is well established in the initial audio and video
   applications that are expected to use RTP, a confidentiality service
   is defined in the next section for use with RTP and RTCP until lower
   layer services are available. The overhead on the protocol for this
   service is low, so the penalty will be minimal if this service is
   obsoleted by lower layer services in the future.

本文关键:rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt
  相关方案
Google
 

本站最佳浏览方式为 分辨率 1024x768 IE 6.0(或更高版本的 IE浏览器)

go top