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Real-Time Protocols
RTP/RTCP/RTSP
Heba El-Dreamly –
Ola nijam -
Real-time multimedia streaming
Real-time multimedia applications
–Video teleconferencing
–Internet Telephony (VoIP)
–Internet audio, video streaming
(A-PDUs)
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
–and delay calculationjitterfor
–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
Ideal Timing – no jitter
00.00.00
00.00.10
00.00.20
00.00.30
00.00.11
00.00.21
00.00.31
Send time
Play time
application
Reality – jitter
00.00.00
00.00.10
00.00.20
00.00.30
00.00.11
Send time
Play time
00.00.21
00.00.25
00.00.35
00.00.37
00.00.47
delay
00.00.40
00.00.41
00.00.51
How does Sequence number and Timestamp help ?
:exampleAudio silent
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.
–Why might this cause problems? silent
Consider audio data
–What should the sender do during silent?
Not send anything
Receiver cannot distinguish between loss
and silent
RTCP
Why not UDP?
UDP offers datagram-like
service
Connectionless , Unreliable ,
Unordered
No Flow , Error , Congestion
control
Port numbers
RTP + UDP usually used for multimedia services
Why not TCP?
TCP does retransmissions  unbounded delays
TCP does not support multicast
TCP congestion control (slow-start) unsuitable for real-time transport
(A-V media)
TCP
(till now)
RTSP
Protocol stack for multimedia services
RTP RTCP
 AVTCORE is to maintain the core
RTP/RTCP specifications
 AVTCORE will provide architectural
guidance for extending the protocols
and
 guidelines for their proper use.
 AVTCORE will coordinate closely with
the Security Area
 while working on maintenance and
enhancements to the SRTP Profile.
When watching video clips on the Internet or listen to
audio clips such as radio and others, as well as review in
the games online, we need to Transmission control
protocol, reception and transfer data in real time or in
other words, the flow of data directly.
RTP: Introduction
The Organization for the development of IETF protocol
called Real-time Transport Protocol or Acronym RTP.
RTP Control Protocol (RTCP)
 RTCP specifies report exchanged between sources and
destinations of multimedia information
 receiver reception report
 sender report
 source description report
 Reports contain statistics such as the number of RTP sent,
number of RTP lost, inter-arrival jitter
 Used by application to modify sender transmission rates .
Actually the RTP protocol consists of two parts, one is the
same protocol and the other part is the RTCP protocol,
where he works alongside. While working RTP protocol to
transfer data and arrange them when they arrive in the
wrong sequence, RTCP organizes this process by
transferring control information, such as the
synchronization of multiple transport channels and
monitor the process used for QoS quantity of information
or the Bandwidth occupied equal to 5%
All RTP sent to an even-numbered UDP port, 2p
All RTCP sent to UDP port 2p+1
All RTP and RTCP are sent to same multicast group
Of the basic things that have been taken into
consideration when designing this protocol is the
support for many formats, including
( H.264, MPEG-4, MJPEG, MPEG)
Real-Time Streaming Protocol (RTSP)
Application layer protocol (default port 554)
Usually runs on RTP for stream & TCP for control
Provides the control channel
Uses out-of-band signaling
Usable for Live broadcasts / multicast
Also known as “Network remote control” for multi-media servers.
web
browser
media
player
Web Server
Web Server/Media server
RTSP Overview
RTSP
pres. desc,streaming commands
RTP/RTCP
audio/video content
Presentation
descriptor
HTTP
presentation descriptor
RTSP Methods
OPTIONS
C S
determine capabilities of server/client
C S
DESCRIBE C S get description of media stream
ANNOUNCE C S announce new session description
SETUP C S create media session
RECORD C S start media recording
PLAY C S start media delivery
PAUSE C S pause media delivery
REDIRECT C S redirection to another server
TEARDOWN C S immediate teardown
SET_PARAMETER C S change server/client parameter
GET_PARAMETER C S read server/client parameter
RTSP Session
media server
RTSP
server
data
source
media player
AV
subsystem
RTSP
client
RTSP OK
RTSP PLAY
RTSP OK
RTP AUDIO
RTP VIDEO
RTSP TEARDOWN
RTSP OK
get UDP port
choose
UDP port
RTSP SETUP
Default port
554
RTCP
TCP
UDP

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RIP RTCP RTSP

  • 2. Real-time multimedia streaming Real-time multimedia applications –Video teleconferencing –Internet Telephony (VoIP) –Internet audio, video streaming (A-PDUs)
  • 3. 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 –and delay calculationjitterfor –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
  • 4. Ideal Timing – no jitter 00.00.00 00.00.10 00.00.20 00.00.30 00.00.11 00.00.21 00.00.31 Send time Play time application
  • 5. Reality – jitter 00.00.00 00.00.10 00.00.20 00.00.30 00.00.11 Send time Play time 00.00.21 00.00.25 00.00.35 00.00.37 00.00.47 delay 00.00.40 00.00.41 00.00.51
  • 6. How does Sequence number and Timestamp help ? :exampleAudio silent 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. –Why might this cause problems? silent Consider audio data –What should the sender do during silent? Not send anything Receiver cannot distinguish between loss and silent
  • 7. RTCP Why not UDP? UDP offers datagram-like service Connectionless , Unreliable , Unordered No Flow , Error , Congestion control Port numbers RTP + UDP usually used for multimedia services Why not TCP? TCP does retransmissions  unbounded delays TCP does not support multicast TCP congestion control (slow-start) unsuitable for real-time transport (A-V media)
  • 8. TCP (till now) RTSP Protocol stack for multimedia services RTP RTCP
  • 9.  AVTCORE is to maintain the core RTP/RTCP specifications  AVTCORE will provide architectural guidance for extending the protocols and  guidelines for their proper use.  AVTCORE will coordinate closely with the Security Area  while working on maintenance and enhancements to the SRTP Profile.
  • 10. When watching video clips on the Internet or listen to audio clips such as radio and others, as well as review in the games online, we need to Transmission control protocol, reception and transfer data in real time or in other words, the flow of data directly. RTP: Introduction The Organization for the development of IETF protocol called Real-time Transport Protocol or Acronym RTP.
  • 11. RTP Control Protocol (RTCP)  RTCP specifies report exchanged between sources and destinations of multimedia information  receiver reception report  sender report  source description report  Reports contain statistics such as the number of RTP sent, number of RTP lost, inter-arrival jitter  Used by application to modify sender transmission rates .
  • 12. Actually the RTP protocol consists of two parts, one is the same protocol and the other part is the RTCP protocol, where he works alongside. While working RTP protocol to transfer data and arrange them when they arrive in the wrong sequence, RTCP organizes this process by transferring control information, such as the synchronization of multiple transport channels and monitor the process used for QoS quantity of information or the Bandwidth occupied equal to 5%
  • 13. All RTP sent to an even-numbered UDP port, 2p All RTCP sent to UDP port 2p+1 All RTP and RTCP are sent to same multicast group Of the basic things that have been taken into consideration when designing this protocol is the support for many formats, including ( H.264, MPEG-4, MJPEG, MPEG)
  • 14.
  • 15. Real-Time Streaming Protocol (RTSP) Application layer protocol (default port 554) Usually runs on RTP for stream & TCP for control Provides the control channel Uses out-of-band signaling Usable for Live broadcasts / multicast Also known as “Network remote control” for multi-media servers.
  • 16. web browser media player Web Server Web Server/Media server RTSP Overview RTSP pres. desc,streaming commands RTP/RTCP audio/video content Presentation descriptor HTTP presentation descriptor
  • 17. RTSP Methods OPTIONS C S determine capabilities of server/client C S DESCRIBE C S get description of media stream ANNOUNCE C S announce new session description SETUP C S create media session RECORD C S start media recording PLAY C S start media delivery PAUSE C S pause media delivery REDIRECT C S redirection to another server TEARDOWN C S immediate teardown SET_PARAMETER C S change server/client parameter GET_PARAMETER C S read server/client parameter
  • 18. RTSP Session media server RTSP server data source media player AV subsystem RTSP client RTSP OK RTSP PLAY RTSP OK RTP AUDIO RTP VIDEO RTSP TEARDOWN RTSP OK get UDP port choose UDP port RTSP SETUP Default port 554 RTCP TCP UDP

Editor's Notes

  1. Version : ويتكون من 2 bits والذي يحدد نسخة البروتوكول. النسخة الحالية المستخدمة هي الإصدار 2.P : مختصر Padding يتكون من 1 بت ويستخدم لمعرفة إذا كان هناك bytes إضافية في نهاية الباكيتالمرسلة. هذه الـ bytes تضاف عندما يكون هناك حجم معين من الباكيت يتطلب وجوده كقالب عند استعمالها في خوارزميات التشفير مثلا.X : مختصر Extension يتكون من 1 بت ويستخدم لمعرفة إذا كان هناك bytes إضافية ملحقة بالهيدر أم لا كما ذكرت سابقا.CC : مختصر CSRC Count يتكون من bits 4 وسيتم ذكر فائدته لاحقا في CSRC.M : مختصر Marker يتكون من 1 بت ويستخدم لتفعيل خاصية تضمين المعلومات الخاصة بحدود الفريم من ضمن الباكيت المرسلة.PT : مختصر Payload Type يتكون من bits 7 ويستخدم لمعرفة صيغة الـPayload  وكيفية ترجمة أجزاءه من قبل الApplication Layer .Sequence Number : يأخذ رقم عشوائي يتكون من bits 16 ومن ثم يزداد بمقدار واحد عند كل إرسال ، وعلى الرغم من استخدامه في معرفة إذا كان هناك فقد للبيانات أو وصولها بترتيب خاطئ عند المستقبل إلا انه عديم الفائدة في هذا البروتوكول حيث يترك الأمر للتطبيق المستخدم لتحديد القرار الذي يتطلب اتخاذه. فمثلا بعض التطبيقات تقوم بعرض آخر فريم تم استلامه بدلا من الفريم المفقود، لهذا فهو يستعمل فقط لمعرفة إذا ما تم فقد بيانات من عدمها كونه يعتمد على بروتوكول UDP في عمله.:Timestamp يتكون من 32 bits ويستعمل لتفعيل إمكانية عرض الملف بـSampling rate معين. SSRC : يتكون من 32 bits ويحمل رقم عشوائي يمثل المصدر المستخدم في المزامنة بين streams.CSRC : يتكون من 32 bits ويحدد مصادر البيانات الموجودة في حقل الـ Payload عندما يتم نقل البيانات من أكثر من مصدر. أما عدد هذه المصادر والذي يكون 15 كحد أقصى فيتم تحميله في حقل CC.
  2. أي نظام سريان المعلومات في الزمن الحقيقي. هذا النظام يُمكن المستخدم من إعطاء اوامر شبيهة بأجهزة الفيديو البيتي مثل "توقف" و"تشغيل" وطلب المعلومات من الخادم في نقطة زمن معطاة.النظام يشبه في شكله وأوامره نظام HTTP, ولكنه يضيف بعض الطلبات الجديدة. أحد الفروقات البارزة بين HTTP وRTSP هي أن الأول يعمل دون حفظ الحالات, أما الآخر فهو يحفظ حالة المستخدم, وقد يستخدم ID للجلسات من اجل تتبع الجلسات, بهذه الطريقة لا حاجة لاتصال TCP ثابت بين الخادم والمستخدم. تُرسل رسائل RTSP بشكل عام من المستخدم إلى الخادم