Transporting Voice by using IP

Chapter 2

The IP Protocol Suite

• IP is a routed protocol for passing data packets

• Other protocols invoke IP for the purpose of getting these data packets from origin to destination

• So IP must work with higher layer protocols for any application to work properly

• Remember the OSI 7-Layer model?

Internet Standards

• The Internet Society : Non-profit body with overall objectives to keep the internet alive and growing

• The Internet Architecture Board (IAB): Technical advisory group of the Internet Society.

• The Internet Engineering Task Force (IETF): Volunteers who cooperate in the development on Internet standards; equipment vendors, network operators, research institutions.

Internet Standards ctd ...

• Internet Engineering Steering Group (IESG): Manages and controls IETF’s activities, can approve a particular specification.

• Internet Assigned Number Authority (IANA): Responsible for unique numbers, parameter values and meanings.

Internet Standards Process

• Begins life as an Internet draft• Once it is considered complete it can be

published as an RFC (Request for comments)• The RFC is given a number and becomes a

draft standard.• To achieve this it must have at least 2

independent successful implementations and interoperability must have been demonstrated.

The IP Datagram Format

Routed vs Routing Protocols• Routed: IP, IPX, Novell IPX, Open Standards Institute

networking protocol, DECnet, Appletalk, Banyan Vines, Xerox Network System (XNS).

• Routing: Routing Information Protocol (RIP and RIP II) Open Shortest Path First (OSPF) Intermediate System to Intermediate System (IS-IS) Interior Gateway Routing Protocol (IGRP) Cisco's Enhanced Interior Gateway Routing Protocol

(EIGRP) Border Gateway Protocol (BGP)

Transmission Control Protocol (TCP)

• Ensures that packets are delivered to destination in sequence

• Primary function is to overcome the limitations of IP through an end-to-end confirmation

• Port Numbers: Is a means of identifying a specific instance of a given application.

• Other header fields?

TCP Header Format

Real-time Transport Protocol

User Datagram Protocol (UDP)

• Passes data from and application to IP to be routed to the far end.

• At the far end it simply passes incoming data from IP to the application.

• Provides no acknowledgement functionality• What happens if a UDP packet is lost?• Checksum simply checks that received data is

error free

UDP Header Format

Voice over UDP, not TCP

• Speed is more important than loss of data• Voice packets are smaller so drop of a few will

not be noticeable in the overall context.• Packet loss of about 5% is generally acceptable• Provided that loss is fairly evenly divided• What happens if they arrive out of sequence?• QOS techniques can involve establishing a set

pattern through the network

Real Time Protocol (RTP)

• A Transport Protocol for Real Time Applications

• Sits on top of UDP• Helps address some of the problems

associated with UDP in terms of packet loss• RTP contains a companion protocol (RTCP)• RTCP provides exchange of messages between

sessions to ensure some sort of reliability

Fast-forward to the Year 2021

• Director of Development for MME, Inc.

VideoConferencing

StreamingAudio Movies ?

You

Common Service

Two Goals of RTP’s Common Service

• General enough to be truly “common”– Who knows what applications are coming?– Throughout history, communication has changed:• Oral (traditions passed between generations)• Written• Visual

• Specific enough to actually be useful

RTP can deliver• Multimedia applications requirements• RTP architecture• RTP details• RTP does meet the requirements

Requirements (1)• Timing– Time-stamping for buffered playback• to minimize jitter

– Synchronization of multiple streams– Dynamic frame boundaries• Video: frame length varies due to compression• Audio: “talkspurts”

Requirements (2)• Network issues– Dealing with packet loss– Dealing with congestion• Even with multicast

– Bandwidth utilization • Minimize header bits

Requirements (3)• Miscellaneous– Interoperability• Encoding• Compression

– ID of source• To whom am I listening?• Useful especially in video-conferencing

Requirements Summary• This is not TCP!– Who cares if we lose a packet or two?

– Who cares if we have jitter?

• Calls for a different protocol...

RTP Architecture“ALF” and “ILP”

• Application-level framing:– The application best knows its own needs– May not ask for retransmission, but for lower resolution

• Integrated Layer Processing– Tightly coupled layers– Keeps data presentation from being the bottleneck

• Gives the app. access to the data ASAP!

RTP: Summary• A very thin protocol– Usually built into application

• No hard QOS guarantees– Designed for soft real-time apps– Depends on underlying network– Can run over ATM

• Two components:– Media(data) transport: RTP – Control: RTCP

RTP Concepts• Port numbers for both RTP and RTCP• Participant IP addresses– Strength is multicast

• Relays– Mixers– Translators

RTP Header Format

RTCP• 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

Mixers• Mixer: An application that enable multiple

media streams from different sources to be combined into one overall RTP stream– Could receive and combine various sources in an

effort to reduce bandwidth

Translators

• Used to manage communications between entities that do not support the same media formats or bit rates: e.g. TDM to STDM– Keeps incoming sources separate– To transform to a lower quality format to

broadcast on lower-speed networks– To send through firewalls

Compression• Can use various types– JPEG– MPEG– H.261

• Provided by application• Negotiated using RTCP

Calculation Round-Trip Time (RTT)

• This is another function of RTCP• Useful metric when measuring voice quality• T1, T2, T3 and T4• RTT = T4 - T3 + T2 - T1 • or T4 - (T3 - T2) - T1

Calculation Jitter

• Jitter is defined as the mean deviation of the difference in packet spacing at the receiver compared to packet spacing at the sender for a pair of packets.

• If Si is timestamp for packet i and Ri is the time of arrival in RTP timestamp units for packet i then for 2 packets i and j the deviation in transmit time D is given by:

• D(i,j) = (Rj-Ri) – (Sj-Si) = (Rj-Sj) – (Ri-Si)

IP Multicast

• An example of this with VoIP is a conference call

• Send a packet to a single destination address associated with all listeners

• 224.0.0.1 All hosts on a local subnet• 224.0.0.2 All routers on a local subnet• 224.0.0.5 All routers supporting OSPF• 224.0.0.9 All routers supporting RIP v2

Summary• Multimedia applications have much different needs

than http or ftp!• RTP meets those needs:

• 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)