Chapter 29 • Applications:Voice And Video Over IP (RTP) • Presenters • Monal Kohli • Koushik Sen
INTRODUCTION • IP was designed to transport data,Ip has successfully carried audio and video since its inception. • Audio is transferred by digitizing analog audio signal to produce a data file,which is transferred and at the other end it is decoded to original analog signal. • AUDIOCLIPS:Short audio recordings. • HARDWARE USED:Coder/Decoder called codec. • Three ways to reduce amount of data generated by digital encoding: a.Take fewer samples per second. b. Use fewer bits to encode each sample. c.Use a digital compression scheme to reduce the size of resulting output.
AUDIO AND VIDEO TRANSMISSION • Audio and video applications are considered as Real time as they obey TIMELY TRANSMISSION and DELIVERY. • To guarantee that a stream is delivered at the same rate that the sender used: Conventional Telephone systems uses ISOCHRONOUS ARCHITECHTURE. 1.. An architecture in which,in which the output is delivered with the same timing as we used to generate input. 2.An IP internet is not isochronous due to :DUPLICATION :DELAY :LACK OF ORDER 3.Variance in delay is called JITTER. 4. Meaningful transmission and reproduction of digitized signals across a network with IP semantics requires :Appending a sequence number to each transmission. :assignment of timestamp to each transmission.
JITTER AND PLAYBACK DISPLAY 1.Receiver handles recreation of signal,after delay by maintaining a playback buffer. 2.PLAYBACK POINT:Predetermined threshold in buffer measured in time units of data to be played. 3.If there is no jitter,new data will arrive at exactly the same rate old data is been extracted and played. DISADVANTAGES:Cannot compensate for datagram loss. Items inserted at a variable rate Items extracted at a fixed rate K
REAL TIME TRANSPORT PROTOCOL 1.RTP:Used to transmit digitized audio or video signals over an IP internet is called RTP. 2.FEATURES:Appends sequence numbers in each packet :Timestamp issued. DISADVANTAGES:Does not ensures timely delivery. 3.It does not ensures timely delivery. 4.Instead each packet begins with a four octet header.
8 FIELDS OF HEADER Refer to fig 29.2 for exact header format. 1.VER:Gives RTP version. 2:P:Specifies whether zero padding follws the payload. 3.X:If the application allows extension,X bit is used to specify if extension is present in packet. 4.CC.:Gives a count of contributing sources. 5.M:Used to mark points in the datastream. 6.PTYPE:Specifies the payload type. 7:SEQUENCE NUM:Specifies a 16 bit sequence number for the packet. 8:TIMESTAMP:Gives the time at which first octet of digitized data was sampled. 9.SYNCHRONIZATION SOURCE IDENTIFIER:Specifies source of a stream. 10:CONTRIBUTING SOURCE ID:Provides the synchronization ID’s of streams that were mixed together.
RTP ENCAPSULATION • RTP IS A TRANSPORT LEVEL PROTOCOL. • Each RTP message is encapsulated in a UDP datagram. ADVANTAGE: CONCURRENCY. 1. RTP does not use a reserved UDP port number. 2.A port is allocated for use with each session,and the remote application must be informed about the port number.
RTP CONTROL PROTOCOL Another aspect of real time transmission is monitoring of the underlying network during the session and providing out of band communication between the endpoints. RTP CONTROL PROTOCOL(RTCP)provides the needed control functionality. RTCP messages are encapsulated in UDP for transmission and are sent using a protocol number one greater than the port number Of the RTP stream to which they pertain.
RTCP OPERATION RTCP uses five basic message types to allow senders and recievers to exchange information about a session. 200:SENDER REPORT 201:RECEIVER REPORT 202:SOURCE DESCRIPTION MESSAGE 203:BYE MESSAGE 204:APPLICATION SPECIFIC MESSAGE 1.BYE MESSAGE: A sender transmits a bye message when shutting down a stream. 2.APPLICATION MESSAGE: Provides an extension of the basic facility to allow the application to define a message type. 3.RECEIVER REPORT:receivers periodically transmit receiver report messages that inform the source about conditions of reception. 4.SENDER REPORT:Senders periodically transmit a sender report message that provides an absolute timestamp. 5.SOURCE DESCRIPTION MESSAGE:Senders also transmit source description messages which provide general information about the user who owns or controls the source.
IP Telephony and Signaling • The use of IP as the foundation for telephone services. • Question - what additional technologies are needed before IP can be used in place of the existing isochronous telephone system? • RTP needed to transfer a digitized signal across an IP internet correctly. • Mechanism needed to establish and terminate calls. • How to make IP internet function like an isochronous network. • SIGNALING SYSTEM: • IP telephony must be compatible with extant telephone standards. • The general approach to interoperability uses a gateway between I phone system and the conventional phone system.
There are 2 proposed Standards for IP telephony • I. ITU has defined a suite of protocols known as H.323 • II. IETF proposed a signaling protocol known as Session Initiation Protocol (SIP). • H.323 Standards: • Specifies how multiple protocols can be combined to form a functional IP telephony system • H.323 defines devices known as gatekeepers • Allows the transfer of data in addition to transmission of real time voice and video.
H.323 relies on four major protocols • PROTOCOL • H.225.0 • H.245 • RTP • T.120 • PURPOSE • Signaling used to establish a call. • Control and feedback during the call • Real time data transfer (sequence and timing) • Exchange of data associated with a call
The Fig.29.5 illustrates relationship among the protocols that comprise H.323 Data Applications Signaling and control Audio/video applications Audio codec Video codec H.225 Registr H.225 Signaling H.245 Control T.120 Data RTCP RTP TCP UDP IP
Session Initiation Protocol (SIP) • IETF proposed an alternative to H.323 which only covers signaling • SIP uses client server interaction • SIP relies on a companion protocol,the Session Description Protocol(SDP) to provide information about a call.
Resource Reservation and Quality of Service • Quality of Service (QoS) refers to the statistical performance guarantees that a network system can make regarding loss,delay,throughput and jitter. • Question - Is guaranteed QoS needed for real time transfer of voice and video over IP?
QoS, Utilization & Capacity • Central issue is utilization • Proponents for QoS mechanism assert that QoS achieve two goals- • Divides the existing resources among more users • Allow the network to run at higher utilization by shaping the traffic from each user. • Major argument against complicated QoS arises from improvements in the performance of underlying networks.
Two schemes that can be used to provide QoS in an IP environment • I).RSVP (Resource Reservation Protocol) • II). Common Open Policy Services (COPS) protocol. • RSVP: An endpoint uses RSVP to request a simplex flow through an IP internet with specified QoS bounds.if routers along the path agree to honor the request,they approve it;otherwise,they deny it.if an application needs QoS in two directions,each endpoint must use RSVP to request a separate flow.
COPS • When an RSVP request arrives,a router must evaluate two aspects-feasibility and policies • The router becomes a client that consult a server known a Policy Decision Point(PDP). • COPS protocol defines the client server interaction between a router and a PDP.
Summary • RTP is used to transfer real time data across an IP network Contains- • I. Sequence Number • II. Media Timestamp • Associated control protocol RTCP used to supply information about sources • Debate over whether QoS guarantees are needed to provide real time. • End points use RSVP to request a flow with specific QoS. • Router uses COPS protocol to verify that request meets policy constraints.