Instructor: Prof. Hall Conducted by: Wayde Anwar Vikas Choong Nathan Nadia

29.1 Introduction This chapter focuses on real-time audio/video transfer over IP networks. This chapter focuses on real-time audio/video transfer over IP networks. It examines the question of how IP can be used to provide commercial telephone service. It examines the question of how IP can be used to provide commercial telephone service. It examines the question of how routers in an IP network can guarantee sufficient service to provide HiQ A/V reproduction. It examines the question of how routers in an IP network can guarantee sufficient service to provide HiQ A/V reproduction.

29.2 Audio Clips and Encoding Standards Simplest digitizing A/D (encoding) -> IP network-> D/A (decoding) Simplest digitizing A/D (encoding) -> IP network-> D/A (decoding) –OK for audio clips, not for interactive b/c of delay introduced –HiQ codec are available (Amplitude overtime to sequence of digits, reconstruct from the digits to waveform). –Standards: based on the tradeoffs b/w quality and reproduction and size of digital representation.

29.2 Audio Clips and Encoding Standards(Continued) –E.g. PCM for phone line (huge file production) –Three ways to reduce the size »Fewer samples: Low quality »Fewer bits: Low quality »Compression: Delay(require fast CPU) good when delay is not important. Produce data at 2.2 kbps

29.3 Audio and Video Transmission and Reproduction AV application are real-time: timely transmission (missing data is skipped). AV application are real-time: timely transmission (missing data is skipped). How can a network guarantee that the stream is delivered at exactly the same rate that the sender used? How can a network guarantee that the stream is delivered at exactly the same rate that the sender used? –Telephone system way: the entire system is engineered(digital circuits included) to deliver output at the same rate of the input even for multiple paths.

29.3 Audio and Video Transmission and Reproduction(cont) –IP network is not isochronous for the delay introduced – vary delay is called jitter. –Additional protocol is needed in addition to IP; »Each packet must have timestamp to tell the sender when to play back. »This is important b/c it tells the receiver to pause when a packet is lost or sender stops encoding.

29.4 Jitter and Playback Delay How can a receiver recreate a signal accurately if the network introduces a jitter? How can a receiver recreate a signal accurately if the network introduces a jitter? –Playback buffer (similar to queue) –How does it work? »The receiver introduces a delay until the buffer is filled with incoming data (Threshold-playback point) – figure (K) is the unit of time of data to be played. »The receiver plays K time units. »If no jitter, datagrams continue to arrive at the same rate, so the buffer is filled with K time units of un-played data

29.4 Jitter and Playback Delay(Cont) »If small delay, playback won’t be affected, the buffer decreases as data are extracted, playback continues for K units, once the delayed datagrams arrive buffer will be refilled. »If a datagram is lost, buffer will be empty, output pauses for time corresponding for the missing data. K is small – needed buffer will be used before delayed data arrive. K is small – needed buffer will be used before delayed data arrive. K is too large – immunity to jitter with noticeable delay (in addition to NW delay) to user. K is too large – immunity to jitter with noticeable delay (in addition to NW delay) to user. »Playback is still used despite disadvantages.

29.5 Real-Time Transport Protocol (RTP) It does not provide timely transmission. Timely manner depends on the underlying system. It does not provide timely transmission. Timely manner depends on the underlying system. It provides: It provides: –Sequence Number –Timestamp RTP does not distinguish b/w types of data; therefore, it does not enforce uniform interpretation of semantics. RTP does not distinguish b/w types of data; therefore, it does not enforce uniform interpretation of semantics. For the receiver to control playback.

29.5 Real-Time Transport Protocol (RTP) (Cont) RTP header provides needed information for interpretation by the receiver: RTP header provides needed information for interpretation by the receiver: –2 bit version (current 2) –16 bit SEQUENCE NUM: first one is randomly chosen. –X-bit is used to identify if the application defines optional header extension b/w RTP header and pay load. –7 bit PTYPE: determines the interpretation of the most remaining header field (Pay Load Type).

29.5 Real-Time Transport Protocol (RTP) (Cont) –P-bit specify whether padding is in effect to the pay load. (Encryption: How data is allocated in blocks). –M-bit used by the application (Marking points – e.g. beginning of video stream) –32-bit TIMESTAMP – affected by the type at which first octet is digitized.

29.6 Streams, Mixing, and Multicasting Key Part to RTP is its support for translation or mixing. Key Part to RTP is its support for translation or mixing. –Translation: changing the encoding of a stream at an intermediate station. –Mixing: receiving streams of data from multiple sources, combining them into a single stream, and sending the results. »Mixers are important to service multiple streams in conferencing.

29.6 Streams, Mixing, and Multicasting(Cont.) The field SYNCHRONIZATION SOURCE IDENTIFIER specifies. Each source must choose a unique identifier. If mixer is enabled, the mixer will be the source of the new stream. The field SYNCHRONIZATION SOURCE IDENTIFIER specifies. Each source must choose a unique identifier. If mixer is enabled, the mixer will be the source of the new stream. The original source is not lost b/c mixer uses CONTRIBUTING SOURCE ID to identify the actual stream source. The original source is not lost b/c mixer uses CONTRIBUTING SOURCE ID to identify the actual stream source. CC-field gives the number of contributing sources. CC-field gives the number of contributing sources.

29.6 Streams, Mixing, and Multicasting(Cont.) RTP works with IP multicasting and mixing especially in multicast environment. RTP works with IP multicasting and mixing especially in multicast environment. For example, in teleconference situation, unicast is cumbersome; however, multicasting will allow multi-users to communicate both ways at the same time. Mixers make this possible by reading several inputs resulting in fewer datagrams. For example, in teleconference situation, unicast is cumbersome; however, multicasting will allow multi-users to communicate both ways at the same time. Mixers make this possible by reading several inputs resulting in fewer datagrams.

29.7 RTP Encapsulation RTP is a transport-level protocol working on the top of UDP. RTP is a transport-level protocol working on the top of UDP. This means that it needs to be encapsulated in UDP before the final encapsulation in IP datagram. This means that it needs to be encapsulated in UDP before the final encapsulation in IP datagram. RTP does not have a reserved port number. Port is allocated for each session, and remote app must inform about port number. RTP prefer even numbers. RTP does not have a reserved port number. Port is allocated for each session, and remote app must inform about port number. RTP prefer even numbers.

29.8 RTP Control Protocol So far, Real-Time transmission has been explained as a protocol allowing reproduction of A/V data. So far, Real-Time transmission has been explained as a protocol allowing reproduction of A/V data. Monitoring the underlying network is as important as the protocol itself during each session, and providing out of band com b/w endpoints. (Adaptive applications). Monitoring the underlying network is as important as the protocol itself during each session, and providing out of band com b/w endpoints. (Adaptive applications). An application may adjust the buffer size, or choose lower band width due to NW cong. An application may adjust the buffer size, or choose lower band width due to NW cong.

29.8 RTCP (Cont.) Out of Band can be used to send information in parallel with real time like caption. Out of Band can be used to send information in parallel with real time like caption. RTP control protocol (RTCP) provides the needed control functionality. RTP control protocol (RTCP) provides the needed control functionality. RTCP: allows senders and receivers to transmit a series of reports one to another that contain additional info about data transferred in addition to NW performance. RTCP: allows senders and receivers to transmit a series of reports one to another that contain additional info about data transferred in addition to NW performance. RTCP is encapsulated in UDP using a port number that is greater than RTP port number. RTCP is encapsulated in UDP using a port number that is greater than RTP port number.

29.9 RTCP Operation Uses 5 basic message type: Sender Report - provides absolute timestamp Sender Report - provides absolute timestamp –Absolute timestamp is essential to synchronize multiple streams –Since RTP require separate stream for each media, transmission of video/audio require 2 streams Receiver Report - Inform source about conditions of reception Receiver Report - Inform source about conditions of reception –allow participating receivers & senders in a session to learn about reception conditions of other receivers

29.9 RTCP Operation –allow receivers to adapt their rate of reporting to avoid using excessive bandwidth & overwhelming the sender Source Desc. Message - general info about user (owns/ control source) Source Desc. Message - general info about user (owns/ control source) –Each message contain 1 section for each outgoing RTP stream Bye Message - Shutting down a stream Bye Message - Shutting down a stream Application Specific Message - extend basic facility to allow application to define message type Application Specific Message - extend basic facility to allow application to define message type

29.10 IP telephony & Signaling Real-time transmission: use of IP as the foundation for telephone service Real-time transmission: use of IP as the foundation for telephone service Researches are investigation 3 components to replace isochronous systems: Researches are investigation 3 components to replace isochronous systems: –RTP is needed to transfer a digitized signal across an IP internet correctly –Mechanism is needed to establish and terminate telephone calls –Researches are exploring ways an IP internet can function like an isochronous network

29.10 IP telephony & Signaling Telephone industry use Signaling : process of establishing a telephone call Telephone industry use Signaling : process of establishing a telephone call Public Switched Telephone Network (PSTN) uses Signaling System 7 (SS7) Public Switched Telephone Network (PSTN) uses Signaling System 7 (SS7) – performs call routing before any audio is sent – handles call forwarding and error conditions

29.10 IP telephony & Signaling Signaling functionality must be available before IP can be used to make calls Signaling functionality must be available before IP can be used to make calls IP telephony must be also compatible with extant telephone standards IP telephony must be also compatible with extant telephone standards Must be possible for IP telephony system to interoperate with the conventional phone system at all levels. Must be possible for IP telephony system to interoperate with the conventional phone system at all levels.

29.10 IP telephony & Signaling The general approach to interoperability uses a gateway between IP & conventional phone system The general approach to interoperability uses a gateway between IP & conventional phone system Standards for IP Telephony: Standards for IP Telephony: –ITU has defined a suite or protocol known as H.323 –IETF has proposed a signaling protocols know as SIP

H.323 Standards Originally created to allow the transmission of voice over local area Originally created to allow the transmission of voice over local area Then it was extended to allow transmission of voice over IP internets Then it was extended to allow transmission of voice over IP internets Specifies how multiple protocols can be combined to form functional IP telephony Specifies how multiple protocols can be combined to form functional IP telephony Defines gateways & gatekeepers : Defines gateways & gatekeepers : –provide a contact point for telephones using IP. – Each IP Telephone must register with a gatekeeper

H.323 Standards H.323 relies on 4 major protocols: H.323 relies on 4 major protocols: –H Signaling used to establish a call –H.224Control and feedback during the call –RTPReal-time data transfer –T.120Exchange of data associated with a call Fig 29.5 illustrates relationship among the H.323 protocols Fig 29.5 illustrates relationship among the H.323 protocols

Session Initiation Protocol (SIP) Covers only signaling, doesn't supply all of H.323 functionality Covers only signaling, doesn't supply all of H.323 functionality Uses client-server interaction, with servers being divided into 2 types: Uses client-server interaction, with servers being divided into 2 types: –user agent server runs in a SIP telephone » assigned an identifier: –intermediate server ; between 2 SIP telephone »handles call set up and call forwarding

Session Initiation Protocol (SIP) SIP relies on Session Description Protocols SDP (companion protocol) SIP relies on Session Description Protocols SDP (companion protocol) SDP important in conference call SDP important in conference call –participants join and leave dynamically SDP specifies media encoding, protocols number and multicast address SDP specifies media encoding, protocols number and multicast address

29.11 Resource Reservation/Quality of Service Quality of Service (QoS) refers to statistical performance guarantees Quality of Service (QoS) refers to statistical performance guarantees –regarding loss, delay, jitter and throughput An isochronous network that meet strict perfomacnce bounds provide QoS An isochronous network that meet strict perfomacnce bounds provide QoS Packet switched network doesn't provide QoS Packet switched network doesn't provide QoS Is QoS needed for real-time transfer of voice & video over IP? Is QoS needed for real-time transfer of voice & video over IP?

29.11 Resource Reservation/Quality of Service Internet send audio but operates without QoS Internet send audio but operates without QoS ATM, derived from telephone system model, provide QoS guarantees ATM, derived from telephone system model, provide QoS guarantees IETF adopted a differentiated services approach IETF adopted a differentiated services approach –divide traffic into separate QoS classes –sacrifice fine grain control for less complex forwarding

29.12 QoS Utilization & Capacity Central issue is utilization Central issue is utilization –a network with 1% utilization: doesn’t need QoS –a network with 1o1% utilization: will fail under any QoS Proponent who argue for QoS assert that QoS mechanism is important because: Proponent who argue for QoS assert that QoS mechanism is important because: –by dividing the existing resources among more users, system become more “fair ”

29.12 QoS Utilization & Capacity –by shaping traffic, the network run at higher utilization without danger of collapse As long as rapid increases in capacity continues, QoS represent cause unnecessary overhead As long as rapid increases in capacity continues, QoS represent cause unnecessary overhead When demand rises more rapidly than capacity, it becomes an economic issue When demand rises more rapidly than capacity, it becomes an economic issue

29.13 RSVP How can IP network provide QoS? How can IP network provide QoS? IETF produced 2 protocols: RSVP & COPS IETF produced 2 protocols: RSVP & COPS QoS cannot be added at the application layer to IP; basic infrastructure must change QoS cannot be added at the application layer to IP; basic infrastructure must change Infrastructure must change: routers must agree to reserve resources Infrastructure must change: routers must agree to reserve resources Endpoints must send a request to spefiicy resources needed before data is sent Endpoints must send a request to spefiicy resources needed before data is sent As datagrams traverse the flow, routers need to monitor (traffic policing) and control traffic forwarding As datagrams traverse the flow, routers need to monitor (traffic policing) and control traffic forwarding

29.13 RSVP Control of queuing is needed: Control of queuing is needed: –router must implement a queuing policy that meets guaranteed bounds on delay –router must smooth packet burst (traffic shaping) RSVP is not a routing protocols; operates before any data is sent and handles reservations request and replies. RSVP is not a routing protocols; operates before any data is sent and handles reservations request and replies. RSVP is unidirectional (simplex); if application needs QoS in two directions, each point must use RSVP to request a separate flow RSVP is unidirectional (simplex); if application needs QoS in two directions, each point must use RSVP to request a separate flow

29.14 COPS When an RSVP arrivers a router must evaluate: When an RSVP arrivers a router must evaluate: –feasibility : a local decision –policies: requires global cooperation IETF architecture uses 2-level model: »when router receiver RSVP request, it becomes a client which consult server :Policy Decision Point (PDP) to determine whether request meets policy constraints »if PDP approves a request, router must operate as Policy Point Point (PEP)to ensure traffic does not exceed the approved policy COPS protocol define the client-server interaction between a router and a PDP COPS protocol define the client-server interaction between a router and a PDP

29.14 COPS Although COPS defines it own message header, the underlying format shares many details with RSVP Although COPS defines it own message header, the underlying format shares many details with RSVP When a router receives an RSVP request: When a router receives an RSVP request: –extract items related to policy –place them in a COPS message –send the result to PDP

Summary Audio data can be encoded in digital form (hardware:codec) Audio data can be encoded in digital form (hardware:codec) Pulse Code Modulation (PCM) produce digital values at 64 Kbps Pulse Code Modulation (PCM) produce digital values at 64 Kbps RTP is used to transfer real-time data across an IP internet. Each message contain : RTP is used to transfer real-time data across an IP internet. Each message contain : –sequence number –a media timestamp

Summary RTCP is used to supply information about sources & allow mixer to combine several streams RTCP is used to supply information about sources & allow mixer to combine several streams Debate continues where Q0S guarantees is needed to provide real-time Debate continues where Q0S guarantees is needed to provide real-time Endpoints use RSVP to request a flow with specific QoS; intermediate routers either approve or deny the request Endpoints use RSVP to request a flow with specific QoS; intermediate routers either approve or deny the request When RSVP request arrives, router use COPS to contact PDP and verify that request meets policy constraints When RSVP request arrives, router use COPS to contact PDP and verify that request meets policy constraints