RTP ( Real-time Transport Protocol )

• Payload-type identification - Indication of what kind of content is being carried


• Sequence numbering - PDU sequence number


• Time stamping - presentation time of the content being carried in the PDU


• Delivery monitoring

• A profile specification document, which defines a set of payload type codes and their mapping to payload formats (e.g., media encodings). A profile may also define extensions or modifications to RTP that are specific to a particular class of applications. Typically an application will operate under only one profile. A profile for audio and video data may be found in the companion RFC .



• Payload format specification documents, which define how a particular payload, such as an audio or video encoding, is to be carried in RTP.

• V (Version)

V is RTP version number. Always set to 2.


• P (Padding)

If set, this packet contains one or more additional padding bytes at the end which are not part of the payload. The last byte of the padding contains a count of how many padding bytes should be ignored. Padding may be needed by some encryption algorithms with fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit.


• X (Extension bit)

When set, the fixed header is followed by exactly one header extension, with a defined format.


• CSRC count

Contains the number of CSRC identifiers that follow the fixed header.


• M (Marker)

The interpretation of the marker is defined by a profile. It is intended to allow significant events such as frame boundaries to be marked in the packet stream. A profile may define additional marker bits or specify that there is no marker bit by changing the number of bits in the payload type field.


• Payload type

Identifies the format of the RTP payload and determines its interpretation by the application. A profile specifies a default static mapping of payload type codes to payload formats. Additional payload type codes may be defined dynamically through non-RTP means.


PT	Name	Type	Clock rate (Hz)	Audio channels
0	PCMU	Audio	8000	1
1	1016	Audio	8000	1
2	G721	Audio	8000	1
3	GSM	Audio	8000	1
4	G723	Audio	8000	1
5	DVI4	Audio	8000	1
6	DVI4	Audio	16000	1
7	LPC	Audio	8000	1
8	PCMA	Audio	8000	1
9	G722	Audio	8000	1
10	L16	Audio	44100	2
11	L16	Audio	44100	1
12	QCELP	Audio	8000	1
13	CN	Audio	8000	1
14	MPA	Audio	90000
15	G728	Audio	8000	1
16	DVI4	Audio	11025	1
17	DVI4	Audio	22050	1
18	G729	Audio	8000	1
19	reserved	Audio
20-24
25	CellB	Video	90000
26	JPEG	Video	90000
27
28	nv	Video	90000
29-30
31	H261	Video	90000
32	MPV	Video	90000
33	MP2T	Audio/Video	90000
34	H263	Video	90000
35-71
72-76	reserved
77-95
96-127	dynamic
dynamic	GSM-HR	Audio	8000	1
dynamic	GSM-EFR	Audio	8000	1
dynamic	L8	Audio	variable	variable
dynamic	RED	Audio
dynamic	VDVI	Audio	variable	1
dynamic	BT656	Video	90000
dynamic	H263-1998	Video	90000
dynamic	MP1S	Video	90000
dynamic	MP2P	Video	90000
dynamic	BMPEG	Video	90000



• Sequence Number

The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random (unpredictable) to make known-plaintext attacks on encryption more difficult, even if the source itself does not encrypt, because the packets may flow through a translator that does.


• Timestamp

The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. The resolution of the clock must be sufficient for the desired synchronization accuracy and for measuring packet arrival jitter (one tick per video frame is typically not sufficient). The clock frequency is dependent on the format of data carried as payload and is specified statically in the profile or payload format specification that defines the format, or may be specified dynamically for payload formats defined through non-RTP means. If RTP packets are generated periodically, the nominal sampling instant as determined from the sampling clock is to be used, not a reading of the system clock. As an example, for fixed-rate audio the timestamp clock would likely increment by one for each sampling period. If an audio application reads blocks covering 160 sampling periods from the input device, the timestamp would be increased by 160 for each such block, regardless of whether the block is transmitted in a packet or dropped as silent.


• SSRC (Synchronization source)

Identifies the synchronization source. The value is chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC. Although the probability of multiple sources choosing the same identifier is low, all RTP implementations must be prepared to detect and resolve collisions. If a source changes its source transport address, it must also choose a new SSRC to avoid being interpreted as a looped source.


• CSRC (Contributing source)

An array of 0 to 15 CSRC elements identifying the contributing sources for the payload contained in this packet. The number of identifiers is given by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC identifiers are inserted by mixers, using the SSRC identifiers of contributing sources. For example, for audio packets the SSRC identifiers of all sources that were mixed together to create packets are listed, allowing correct talker indication at the receiver.

• Simple Multicast Audio Conference

A working group of the IETF meets to discuss the latest protocol document, 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, 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 when it leaves the conference.


• Audio and Video Conference

If both audio and video media are used in a conference, they are transmitted as separate RTP sessions. That is, separate RTP and 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. 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.


• Mixers and Translators

So far, we have assumed that all sites want to receive media data in the same format. However, this may not always be appropriate. Consider the case where participants in one area are connected through a low-speed link to the majority of the conference participants who enjoy high-speed network access. Instead of forcing everyone to use a lower-bandwidth, reduced-quality audio encoding, an RTP-level relay called a mixer may be placed near the low-bandwidth area. This mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packet stream across the low-speed link. These packets might be unicast to a single recipient or multicast on a different address to multiple recipients. The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that correct talker indication can be provided at the receivers.

Some of the intended participants in the audio conference may be connected with high bandwidth links but might not be directly reachable via IP multicast. For example, they might be behind an application-level firewall that will not let any IP packets pass. For these sites, mixing may not be necessary, in which case another type of RTP-level relay called a translator may be used. Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site's internal network.

Mixers and translators may be designed for a variety of purposes. An example is a video mixer that scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene. Other examples of translation include the connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II, or the packet-by-packet encoding translation of video streams from individual sources without resynchronization or mixing.


• Layered Encodings

Multimedia applications should be able to adjust the transmission rate to match the capacity of the receiver or to adapt to network congestion. Many implementations place the responsibility of rate-adaptivity at the source. This does not work well with multicast transmission because of the conflicting bandwidth requirements of heterogeneous receivers. The result is often a least-common denominator scenario, where the smallest pipe in the network mesh dictates the quality and fidelity of the overall live multimedia "broadcast".

Instead, responsibility for rate-adaptation can be placed at the receivers by combining a layered encoding with a layered transmission system. In the context of RTP over IP multicast, the source can stripe the progressive layers of a hierarchically represented signal across multiple RTP sessions each carried on its own multicast group. Receivers can then adapt to network heterogeneity and control their reception bandwidth by joining only the appropriate subset of the multicast groups.

■ Parent layer
■ Child layer

UDP VINES

RTP

Audio Video Audeo/Video Dynamic