Transporting Voice by using IP
-
Upload
dierdra-lysaght -
Category
Documents
-
view
22 -
download
2
description
Transcript of Transporting Voice by using IP
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.
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?
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
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
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)