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RTP

RTP is a protocol for delivering audio and video over IP networks. It provides mechanisms for sequencing packets, time stamping media, identifying codecs, and detecting packet loss. RTP is usually used with UDP for real-time multimedia applications. A related protocol, RTCP, monitors transmission quality and allows participants to identify themselves. Together, RTP and RTCP enable applications to stream audio and video with mechanisms for synchronization, adaptation to network conditions, and interoperability between systems.

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49 views

RTP

RTP is a protocol for delivering audio and video over IP networks. It provides mechanisms for sequencing packets, time stamping media, identifying codecs, and detecting packet loss. RTP is usually used with UDP for real-time multimedia applications. A related protocol, RTCP, monitors transmission quality and allows participants to identify themselves. Together, RTP and RTCP enable applications to stream audio and video with mechanisms for synchronization, adaptation to network conditions, and interoperability between systems.

Uploaded by

tfxp
Copyright
© Attribution Non-Commercial (BY-NC)
Available Formats
Download as PPT, PDF, TXT or read online on Scribd
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 Multimedia applications’ requirements

 RTP architecture

 RTP details

 RTCP details

 Mixer and Translators

 Applications of RTP

 Summary
Real-time multimedia applications
Video teleconferencing
Internet Telephony (VoIP)
Internet audio, video streaming
Streaming performance requirements
– Sequencing
– to report PDU loss
– to report PDU reordering
– to perform out-of-order decoding

– Time stamping and Buffering


– for play out
– for jitter and delay calculation

– Payload type identification


– for media interpretation

– Error concealment –covers up errors from lost PDU by using redundancy in


most-adjacent-frame

– Quality of Service (QoS) feedback – from receiver to sender for operation


adjustment

– Rate control –sender reduces sending rate adaptively to network congestion


Jitter
Playback buffer
At time 00:00:18

At time 00:00:28

At time 00:00:38
How does Sequence number and Timestamp help ?

Audio silence example: Seq no.1


, Tmpst 1
00
Seq no.2
, Tmpst 2
Consider audio data 00
Seq no.3
– What should the sender do during silence? , Tmpst 3
00

receiver
Not send anything

sender
– Why might this cause problems? silence
Receiver cannot distinguish between loss and
silence Seq no.4
, Tmpst 6
00
Seq no.5
, Tmpst 7
00

Solution:

– After receiving no PDUs for a while, next PDU received


at the receiver will reflect a big jump in timestamp, but
have the correct next seq. no. Thus, receiver knows what
happened.
Support from Transport Layers

TCP is not used because:


• TCP does retransmissions  unbounded delays
• No provision for time stamping
• TCP does not support multicast
• TCP congestion control (slow-start) unsuitable for real-time
transport

RTP + UDP usually used for multimedia services


 Minimized jitter
 Synchronized sources

 Dynamic, payload-specific frame length

 Adaptation in the face of congestion

 Interoperability

 Effective use of bandwidth

 Support for video-conferencing (multicast, IDs)


 Developed by the Audio-Video Transport Working
Group of the IETF

 First published as RFC 1889 in the year 1996

 Superseded by RFC 3550 in the year 2003


 End-to-end delivery services for applications transmitting real-
time data, such as audio and video
 Payload type identification
 Sequence numbering
 Time stamping
 Delivery monitoring
 Lightweight

 Flexible

 protocol-neutral

 scalable

 separate control/data
 secure
RTP Session

 RTP session is sending and receiving of RTP data by a group


of participants
 For each participant, a session is a pair of transport
addresses used to communicate with the group
 If multiple media types are communicated by the group, the
transmission of each medium constitutes a session.
Is there an extension
Is it padded? number of contributing sources
mpeg?, 8-bit PCM ? etc.
header?
version
Is there any loss?
source of this packet when this packet should be played

The RTP header.


 ID of sender
 Provides various reports for use in:
 QoS and congestion control
 so an app can change resolution or compression
strategies
 Session size and scaling
 conferencing
RTCP Messages Types RTCP Packet
 Accommodate participant network resources
 Mixer – Low Bandwidth
 Mixer – Combining media streams
 Translator – Forward RTP packets to private networks
 Define media data formats or encodings
 Need media specific profiles

 Handle connection setups or tear-downs


 Need other protocols like SIP or H.323

 Handle resource reservation


 Need other protocols like RSVP

 Guarantee timely delivery or Quality of Service


 However, it does provide necessary data to application to order packets and adjust
signal quality
Real-time Transport Protocol 12/07/21

1. Call connection established


2. Audio sampled at 20ms durations
3. Each data chunk is packaged with an RTP header
4. RTP packet is wrapped around UDP packet
5. Sent through network
6. Receiver receives and parses RTP header
7. Based on payload type, application reconstructs audio
stream in 20ms chunks

19
Real-time Transport Protocol 12/07/21

 Audio and Video has


separate RTP/RTCP sessions
 Receiver can choose media type
to use
 Encodings can change
independently
 Network resource allocations
per media basis

20
THANK
YOU!!!

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