A working group of the IETF meets to discuss the latest protocol
draft, using the IP multicast services of the Internet for voice
communications. Through some allocation mechanism the working group
chair obtains a multicast group address and pair of ports. One port
is used for audio data, and the other is used for control (RTCP)
packets. This address and port information is distributed to the
intended participants. If privacy is desired, the data and control
packets may be encrypted as specified in Section 9.1, in which case
an encryption key must also be generated and distributed. The exact
details of these allocation and distribution mechanisms are beyond
the scope of RTP.
The audio conferencing application used by each conference
participant sends audio data in small chunks of, say, 20 ms duration.
Each chunk of audio data is preceded by an RTP header; RTP header and
data are in turn contained in a UDP packet. The RTP header indicates
what type of audio encoding (such as PCM, ADPCM or LPC) is contained
in each packet so that senders can change the encoding during a
conference, for example, to accommodate a new participant that is
connected through a low-bandwidth link or react to indications of
network congestion.
The Internet, like other packet networks, occasionally loses and
reorders packets and delays them by variable amounts of time. To cope
with these impairments, the RTP header contains timing information
and a sequence number that allow the receivers to reconstruct the
timing produced by the source, so that in this example, chunks of
audio are contiguously played out the speaker every 20 ms. This
timing reconstruction is performed separately for each source of RTP
packets in the conference. The sequence number can also be used by
the receiver to estimate how many packets are being lost.
Since members of the working group join and leave during the
conference, it is useful to know who is participating at any moment
and how well they are receiving the audio data. For that purpose,
Schulzrinne, et al Standards Track [Page 5]
RFC 1889 RTP January 1996
each instance of the audio application in the conference periodically
multicasts a reception report plus the name of its user on the RTCP
(control) port. The reception report indicates how well the current
speaker is being received and may be used to control adaptive
encodings. In addition to the user name, other identifying
information may also be included subject to control bandwidth limits.
A site sends the RTCP BYE packet (Section 6.5) when it leaves the
conference.
2.2 Audio and Video Conference
If both audio and video media are used in a conference, they are
transmitted as separate RTP sessions RTCP packets are transmitted for
each medium using two different UDP port pairs and/or multicast
addresses. There is no direct coupling at the RTP level between the
audio and video sessions, except that a user participating in both
sessions should use the same distinguished (canonical) name in the
RTCP packets for both so that the sessions can be associated.
One motivation for this separation is to allow some participants in
the conference to receive only one medium if they choose. Further
explanation is given in Section 5.2. Despite the separation,
synchronized playback of a source's audio and video can be achieved
using timing information carried in the RTCP packets for both
sessions.
2.3 Mixers and Translators
