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Quality of Service For Voip
Quality of Service For Voip
Dr. Danny Tsang Department of Electronic & Computer Engineering Hong Kong University of Science and Technology
Quality of Service
QoS
network provides application with level of
Quality of Service
A B
packets queueing (delay) free (available) buffers: arriving packets dropped (loss) if no free buffers
Quality of Service
2. queueing
time waiting at output link for transmission depends on congestion level of router
A B
transmission
propagation
nodal processing
queueing
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A B
transmission
nodal processing
queueing
Quality of Service
Nodal delay
d nodal d proc d queue d trans d prop
dproc = processing delay
depends on congestion
= L/R, significant for low-speed links
Quality of Service
rate
Quality of Service
source to router along end-end Internet path towards destination. For all i:
sends three packets that will reach router i on path towards destination router i will return packets to sender sender times interval between transmission and reply.
3 probes 3 probes
3 probes
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Packet loss
queue (aka buffer) preceding link in buffer has
finite capacity when packet arrives to full queue, packet is dropped (aka lost) lost packet may be retransmitted by previous node, by source end system, or not retransmitted at all
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13
MM Networking Applications
Classes of MM applications: 1) Streaming stored audio and video 2) Streaming live audio and video 3) Real-time interactive audio and video Fundamental characteristics: Typically delay sensitive
Jitter is the variability of packet delays within the same packet stream
infrequent losses cause minor glitches Antithesis of data, which are loss intolerant but delay tolerant.
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Streaming: media stored at source transmitted to client streaming: client playout begins before all data has arrived
timing constraint for still-to-be
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1. video recorded
2. video sent
network delay
playing out early part of video, while server still sending later part of video
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pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect OK RTSP often used (more later) timing constraint for still-to-be transmitted data: in time for playout
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video conference, distributed interactive worlds end-end delay requirements: audio: < 150 msec good, < 400 msec OK
includes application-level (packetization) and network delays higher delays noticeable, impair interactivity
applications: IP telephony,
session initialization
how does callee advertise its IP address, port number, encoding algorithms? Quality of Service
19
But you said multimedia apps requires ? QoS and level of performance to be ? ? effective! ?
Todays Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss
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Differentiated services philosophy: Fewer changes to Internet infrastructure, yet provide 1st and 2nd class service.
application layer
at constant rate
Example: 8,000
samples/sec, 256 quantized values --> 64,000 bps Receiver converts it back to analog signal:
i.e., rounded
represented by bits
Example rates CD: 1.411 Mbps MP3: 96, 128, 160 kbps Internet telephony: 5.3 - 13 kbps
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e.g. 24 images/sec
Examples: MPEG 1 (CD-ROM) 1.5 Mbps MPEG2 (DVD) 3-6 Mbps MPEG4 (often used in Internet, < 1 Mbps) Research: Layered (scalable) video
spatial temporal
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periods.
application-layer header added to each chunk. Chunk+header encapsulated into UDP segment.
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Delay Jitter Packet loss Packet mis-order Available bandwidth Network design Endpoint audio characteristics (sound card, microphone, earpiece, etc.)
Transcoding Echo Silence suppression Duplex Codec selection Router and data-switch configuration Wan protocols QoS/CoS policy Encryption Decryption
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Delay
Time cost to traverse the networks
Voice calls are real-time, full-duplex
communications, end-to-end delay of packets can have severe impacts on usability of the VoIP applications. Contributed by
Propagation delay
caused by the characteristics of the speed of light traveling via a fiber-optic-based or copper-based medium the devices that handle voice information switch, routers, firewall, etc. Time cost to generate a voice packet the DSP generates a frame every 10 milliseconds. Two of these frames are then placed within one voice packet; the packet delay is therefore 20 milliseconds
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31
milliseconds
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Citizens Band (CB) Radio Service is a private two-way voice communication service for use in personal and business activities of the general public. Its communications range is from one to five miles.
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400ms+
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Jitter
Packets have varying transmission times Variation between when a voice packet is expected to be
received and when it actually is received (variations in delay ) Occur when packets get held up in queues because of congestion within the internet Causing a discontinuity in the real-time voice stream Approaches to compensate
Jitter buffer buffer packets with a specified period to smooth packet flow
To increase the size of the jitter buffer However, jitter buffer increases overall delay
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IP datagram lost due to network congestion (router buffer overflow) Due to network outages, network re-routing, etc. delays: processing, queueing in network; end-system (sender, receiver) delays typical maximum tolerable delay: 400 ms
20%, the audio quality of VoIP is degraded beyond Quality of Service usefulness (RFC3714)
36
Transcoding
Convert a voice signal from analog to
digital or digital to analog Calls may experience multiple transcoding when routed with multiple voice coders Result in quality degradation
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Path reservation RSVP Scheduling Priority Queuing, Weighted Fair Queuing, Class-based Queuing
Other elements
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Bandwidth Tradeoffs
Encode and compress
data for transport across the Internet Compression techniques developed for
Comparison Standard Bandwidth (Compressed voice rate) Complexity (CPU usage) Voice quality Digitizing delay MOS, Mean Opinion Score
telephony and voice packet are standardized by the ITU-T in its Gseries recommendations
ITU-T Recommendation P.800 Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 Toll quality, 4.0 or higher
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Audio compression standards Jitter mitigation Loss recovery Echo control Silence Suppression
Voice activity detection (VAD), comfort noise generation (CNG)
MPLS
Virtual LANs (802.1p/q)
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Codec Selection
Depending on The bandwidth availability Acceptable voice quality
Standard G.711 G.729 G.723.1 Coding Type PCM CS-ACELP ACELP Bit Rate (Kbps) 64 8 6.3 Quality rate 4.3 4.0 3.8
MP-MLQ
5.3
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Audio codecs
G.711 Raw telephone audio PCM, Pulse-Code Modulation Highest bandwidth consumption (64 kbps )
SPEEX
open-source (www.speex.org) excellent bandwidth (2 to 44 kbps) excellent quality high CPU usage
GSM G.729 cell phone audio compression low bandwidth usage, low CPU usage, good quality very low bandwidth, low cpu usage broad support from handset manufacturers medium quality Sensitive to packet loss (resilient to bit errors instead of packet loss)
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iLBC
Speech codec developed for robust voice communication over
IP (www.ilbcfreeware.org)) Designed for narrow band speech, with a sampling rate of 8kHz Treat each packet independently from all other packets, ideal for packet communications Graceful speech quality degradation with increasing severity of IP packet loss and /or delay Bitrate 13.33 kbps (399 bits, packetized in 50 bytes) for the frame size of 30 ms and 15.2 kbps (303 bits, packetized in 38 bytes) for the frame size of 20 ms Basic quality higher then G.729A, high robustness to packet loss Computational complexity in a range of G.729A Royalty Free Codec
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44
Audio compression standards Jitter mitigation Loss recovery Echo control Silence Suppression
Voice activity detection (VAD), comfort noise generation (CNG)
MPLS
Virtual LANs (802.1p/q)
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Delay Jitter
constant bit rate transmission client reception
buffered data
time
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msecs after chunk was generated. chunk has time stamp t: play out chunk at t+q . chunk arrives after t+q: data arrives too late for playout, data lost Tradeoff for q: large q: less packet loss small q: better interactive experience
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First packet received at time r First playout schedule: begins at p Second playout schedule: begins at p
packets
loss
playout schedule p' - r playout schedule p-r
time
r p p'
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Estimate network delay, adjust playout delay at beginning of each talk spurt. Silent periods compressed and elongated. Chunks still played out every 20 msec during talk spurt.
t i timestamp of the ith packet ri the time packet i is received by receiver p i the time packet i is played at receiver ri t i network delay for ith packet d i estimate of average network delay after receiving ith packet
d i (1 u )d i 1 u( ri ti )
where u is a fixed constant (e.g., u = .01).
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vi (1 u )vi 1 u | ri ti d i |
The estimates di and vi are calculated for every received packet, although they are only used at the beginning of a talk spurt. For first packet in talk spurt, playout time is:
pi ti d i Kvi
where K is a positive constant. Remaining packets in talkspurt are played out periodically
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difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.
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Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study
Jitter mitigation Loss recovery Echo control Voice activity detection (VAD), comfort noise generation (CNG)
Beyond Best Effort Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP MPLS Virtual LANs (802.1p/q)
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be fixed to the time to receive all n+1 packets Tradeoff: increase n, less bandwidth waste increase n, longer playout delay increase n, higher probability that 2 or more chunks will be lost
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receiver can conceal the loss. Can also append (n-1)st and (n-2)nd low-bit rate chunk
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Interleaving chunks are broken up into smaller units for example, 4 5 msec units per chunk Packet contains small units from different chunks
most of every chunk has no redundancy overhead but adds to playout delay
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Loss concealment
There are various techniques for loss concealment
(i.e., hiding losses), such as those used in the Robust Audio Tool (RAT):
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for delay server side matches stream bandwidth to available client-to-server path bandwidth
error recovery (on top of UDP) FEC, interleaving retransmissions, time permitting conceal errors: repeat nearby data
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Audio compression standards Jitter mitigation Loss recovery Echo control Silence Suppression
Voice activity detection (VAD), comfort noise generation (CNG)
MPLS
Virtual LANs (802.1p/q)
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Echo in VoIP
A common impairment of digital telephone connections is
Echo.
reflected back to its sender interrupting communication. Only speaker hears, receiver does not. A beneficial echo is called Sidetone. It is an effect and applied technique allowing the speaker to hear their own voice in the handsets speaker, convincing themselves their voice is being heard. The benefits of Sidetone are felt when a speaker hears their own voice at a low volume within between 5 and 25 milliseconds (ms). Call quality begins to be is impaired when a speaker hears their speech repeated after approximately 30 ms and more. Types of Echo
Electrical Echo, also known as line, hybrid or network echo Acoustic Echo
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Electrical Echo
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Acoustic Echo
Acoustic Echo occurs in both PSTN and digital networks. Acoustic echo results from the following sources:
Handset use and design (see the above figure): a loudspeaker and a microphone are placed such that the microphone picks up the signal radiated by the loudspeaker and its reflections at the borders of the enclosure. Voice encoding and decoding devices (codecs): an unsuppressed acoustic or electrical echo made worse by digital encoding.
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Echo suppressors
Eliminating Echo
An echo suppressor detects human speech coming from one end of a connection, and suppresses all signals going the other way. An echo suppressor is toggled by a voice recognition circuit. They can typically trip within 5 ms to block a reflected signal. This technique results in a half-duplex channel. This half duplex operation is not noticeable in voice communication, but can adversely effect data communications. Echo suppressors are found on the PSTN installed by Inter-Exchange Carriers (IXCs). An echo canceller is a computer-based device that samples a call and profiles it. An echo will violate the profile. When an echo is detected it simulates the echo, estimates its magnitude, and then subtracts it from the audio signal it is sampling. Echo cancellers are found in modern digital networks at their junction with other networks.
Echo cancellers
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Echo cancellers
Have varying amounts of memory Compare received voice with current
patterns, cancel if match Fail if delay is larger than that the echo canceller memory can afford, which is called echo trail Built into the low bit-rate CODECs and are operated on DSP
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networks. They are less expensive than the analog technology of echo suppressors, and they eliminate in-band signaling requirements. They are also digital. Echo cancellers are typically based on off-the-self Digital Signal Processors (DSPs) that are inexpensive and easy to program.
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DSP solutions: MHz; Memory; Device choice Custom processors: Die size; Technology choice Lower the better
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The G.168 specification does not specify any one design or even recommend a particular tail length. Compliance with G.168 test requirements is necessary but not sufficient.
Only performance requirements laid down by ITU.
Adaptation control, non-linear processor, comfort noise injection, V.25 tone detection, etc. are non-trivial in terms of design and /or resource requirements.
Audio compression standards Jitter mitigation Loss recovery Echo control Silence Suppression
Voice activity detection (VAD), comfort noise generation (CNG)
MPLS
Virtual LANs (802.1p/q)
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Speech Characteristics
silence and speech. VAD algorithms take recourse to some form of speech pattern classification to differentiate between voice and silence periods.
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Silence Suppression
Packets of silence are suppressed to save
Monitor the received signal for voice activity Comfort Noise Generation (CNG).
locally generated white noise make call appear normally connected to both parties when no packets received in silence period
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active or inactive Silence suppression software monitors the received signal for voice activity. Comfort Noise Generation (CNG) - analysis of background noise parameters (encoder) and synthesis of comfort noise (decoder) When no activity is detected for the configured period of time the software informs the Packet Voice Protocol. The encoder output is stopped for bandwidth savings Relay this information to the remote end for comfort noise generation Discontinuous Transmission (DTX) - update background noise parameters Some VADs can cause voice clipping and can result in poor voice quality.
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Speech Encoder
Output
VAD Algorithm
Encoder
Decoder
that can be exploited to give consistent judgment in classifying segments of the signal into silent or voiced segments. Adaptability to Changing Background Noise: Adapting to non-stationary background noise improves robustness, especially in wireless telephony where the user is mobile. Low Computational Complexity: Internet telephony is a real-time application. Therefore the complexity of VAD algorithm must be low to suit real-time applications (not more than one packet time). Toll quality voice reproduction. Saving in bandwidth to be maximized.
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VAD Standards
The latest ITU-T VAD standard is Rec. G.729
Annex B, developed for fixed telephony and multimedia communications. More recently the ETSI has standardized two VADs (options 1 and 2) for the adaptive multirate (AMR) codec developed for thirdgeneration mobile communication systems.
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example algorithm (described in Appendix) equivalent to G.711 without comfort noise flexible for use with any codec guidelines for use in Appendix
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Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study
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Principle 1 packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly
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marking and policing at network edge: similar to ATM UNI (User Network Interface)
fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesnt use
its allocation
Principle 4 Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs
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Evolution of IP-Services
Initially All users were treated equally: no privileges Eventually Different types of usage need different treatment inside the network Evolving concepts (IntServ, DiffServ) Priority by type of services Priority by reservation
Per flow (RSVP) Per aggregate of flows (DiffServ)
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IP Service Classes
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Shaping
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Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study
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arrival to queue
real-world example? discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly
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class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc.. Real world example?
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Policing Mechanisms
Goal: limit traffic to not exceed declared parameters
Three common-used criteria:
Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 15000
ppm peak rate
crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average!
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Policing Mechanisms
Token Bucket: limit input to specified Burst Size
and Average Rate.
full
over interval of length t: number of packets admitted less than or equal to (r t + b).
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arriving
traffic
WFQ
per-flow rate, R
D = b/R max
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MPLS
Virtual LANs (802.1p/q)
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networks for individual application sessions resource reservation: routers maintain state info (a la VC) of allocated resources, QoS reqs admit/deny new call setup requests:
Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?
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request/ reply
Call Admission
Arriving session must :
declare its QOS requirement
defines the QOS being requested characterize traffic it will send into network T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and Tspec to routers (where reservation is required) RSVP
R-spec:
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leaky-bucket-policed source simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988]
arriving traffic token rate, r bucket size, b
approximating the QoS that same flow would receive from an unloaded network element."
WFQ
per-flow rate, R
D = b/R max
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Diffserv approach: simple functions in network core, relatively complex functions at edge routers (or hosts) Dot define define service classes, provide functional components to build service classes
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Diffserv Architecture
Edge router:
per-flow traffic management marks packets as in-profile
r marking b
and out-profile
Core router:
per class traffic management buffering and scheduling based
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Rate A
B
User packets
differently intra-class marking: conforming portion of flow marked differently than non-conforming one
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IPv4, and Traffic Class in IPv6 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive 2 bits are currently unused
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102
Forwarding (PHB)
PHB result in a different observable (measurable)
forwarding performance behavior PHB does not specify what mechanisms to use to ensure required PHB performance behavior Examples:
Class A gets x% of outgoing link bandwidth over time intervals of a specified length Class A packets leave first before packets from class B
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Forwarding (PHB)
PHBs being developed:
Expedited Forwarding: pkt departure rate of a
Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions
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Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study
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path (end system, routers) for QoS for multimedia applications RSVP: Resource Reservation Protocol [RFC 2205]
allow users to communicate requirements to network in robust and efficient way. i.e., signaling !
5.
6.
accommodate heterogeneous receivers (different bandwidth along paths) accommodate different applications with different resource requirements make multicast a first class service, with adaptation to multicast group membership leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes control protocol overhead to grow (at worst) linear in # receivers modular design for heterogeneous underlying technologies
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rather: a mechanism for communicating needs thats the job of routing protocols signaling decoupled from routing separation of control (signaling) and data (forwarding) planes
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unicast destination, or multicast group flowspec: bandwidth requirements spec. filter flag: if yes, record identities of upstream senders (to allow packets filtering by source) previous hop: upstream router/host ID refresh time: time until this info times out path message: communicates sender info, and reversepath-to-sender routing info later upstream forwarding of receiver reservations
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address:
H3 R2 R3
H4
m1:
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
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in routers
m1:
in L7 out L3 L4
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
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tables
m1:
in L3 L4 L7 out L3 L4 L7
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
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115
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
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m1: L7 L6 L6(b) L7
in L3 out L3
L4 L4
L7 L7(b)
H2
L2
b b L1
R1
b L6
H1
R2
L5
b L7
R3
L3 b L4
H3
H4
H5
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b already reserved on L6
m1: in L3 out L3 L4 L4 L7 L7(b)
H2
b L2
b b b L1
H3
R1
b L6
H1
R2
L5
b L7
R3
H4
H5
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H2
b L2
b b b L1
H3
R1
b L6
H1
R2
L5
b L7
R3
H4
H5
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RSVP: example 2
H1, H4 are only senders send path messages as before, indicating filtered reservation Routers store upstream senders for each upstream link H2 will want to receive from H4 (only)
H2
L2
H3
L3 L1
R1
L6
R2
L7
R3
L4
H4
H1
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RSVP: example 2
H1, H4 are only senders send path messages as before, indicating filtered reservation
in
in ; H4-via-R2 ) ) )
L4, L7 ; H1-via-R3 ) ) )
H2
L2
H3 R2
L1 L3 L7 L6, L7 ) )
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R1
L6 in
R3
L4
H4
H1
RSVP: example 2
receiver H2 sends reservation message for source H4
at bandwidth b
in
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H4
H1
L6, L7
RSVP: soft-state
senders periodically resend path msgs to refresh (maintain) state
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H4
H1
L6, L7
RSVP: soft-state
suppose H4 (sender) leaves without performing teardown eventually state in routers will timeout and disappear!
in
in ;H4-via-R2 (b)) ) )
L4, L7 )
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H1
gone H4 fishing!
L6, L7
teardown
to be made to a receiver from a sender who has joined since receivers last reservation refresh
E.g., in previous example, H1 is only receiver, H3 only sender. Path/reservation messages complete, data flows H4 joins as sender, nothing happens until H3 refreshes reservation, causing R3 to forward reservation to H4, which allocates bandwidth
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RSVP: reflections
multicast as a first class service receiver-oriented reservations use of soft-state
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Virtualization of networks
Virtualization of resources: a powerful abstraction in systems engineering: computing examples: virtual memory, virtual devices Virtual machines: e.g., java IBM VM os from 1960s/70s layering of abstractions: dont sweat the details of the lower layer, only deal with lower layers abstractly
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differing in:
addressing conventions packet formats error recovery routing
ARPAnet
"A Protocol for Packet Network Intercommunication", V. Cerf, R. Kahn, IEEE Transactions on Communications, May, 1974, pp. 637-648.
satellite net
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gateway
ARPAnet
satellite net
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internetwork layer underlying local network technology cable satellite 56K telephone modem today: ATM, MPLS invisible at internetwork layer. Looks like a link layer technology to IP!
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just like dialup link is really part of separate network (telephone network)
borrowing ideas from Virtual Circuit (VC) approach but IP datagram still keeps IP address!
MPLS header
IP header
label
20
Exp S TTL
3 1 5
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signaling protocol needed to set up forwarding RSVP-TE forwarding possible along paths that IP alone would not allow (e.g., source-specific routing) !! use MPLS for traffic engineering must co-exist with IP-only routers
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10 12 8
A D A
0 0 1
in label
out interface
10 12
6 9
A D
1 0
R6
0
1 0 1
R4 R5
R3
0
0 in label outR1 label dest
A
out interface
R2
in label out label dest out interface
0
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135
to external network
IP subnet
switch
switch
switch
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their own IP networks (typically subnets) for network management, performance, security and other policy reasons. Users on LANs should be grouped by their community of interest (sales dept., engineering, accounting), not by their location in the building. However, users within a single community of interest are rarely located in the same part of a building. Ethernet switches are easy, routers are hard. Given all the above, how can we separate users via switches? What are the benefits to users and network administrators?
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to external network
IP subnet
switch switch
switch
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138
Virtual LANs
Switches are easy, routers are hard. Given
this, how can we separate users via switches? What are the benefits? Virtual LANs provide separate collision and broadcast domains for groups of users. Users are assigned to one or more VLANs automatically or via a management system. VLANs can span multiple switches and sites How do users on different VLANs talk to each other?
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Virtual LANs
VLANs are LAN segments (in the classic sense of the
word) that can span multiple ethernet switches. VLANs provide separate collision and broadcast domains for each group of users. Users are assigned to one or more VLANs automatically or via a management system. Potential advantages of VLANs include:
Better isolation between groups of users: however it is incorrect to think that VLANs significantly improve network security. Improved performance: the specific LAN performance requirements of each group can be met more easily. Improved performance: VLANs provide multiple broadcast domains Provides for more sophisticated network administration
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IEEE 802.1p
Supplement to MAC Bridges: Traffic Class
Expediting and Dynamic Multicast Filtering, IEEE P802.1p/D6. Extended encapsulation (802.1Q). Method to define relative priority of frames (user_priority). IEEE 802.1p support in LAN switches would provide transmission servicing based on relative priority indicated in each frame (delay indication).
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Tagging 802.1Q
6 bytes 6 bytes 4 bytes 2 bytes Up to 1500 bytes 4 bytes
Destination address
Source address
802.1Q Tag
Type field
Data field
CRC
6 bytes
6 bytes
4 bytes
2 bytes
Up to 1496 bytes
4 bytes
Destination address
Source address
802.1Q Tag
Length field
Data field
CRC
Octet 1
Octet 2 1 2 3 4
Octet 3 5 6 C FI 7 8
Octet 4
but they are intertwined by the technology. VLANs are identified by a 12 bit VLAN Identifier. Frame priority is marked by a 3 bit field, 0 to 7. This is known as Class of Service. Switches can and do, write or re-write, the priority field based on:
Port on the frame was received MAC address of the sending station Protocol IP, IPX, etc. IP Precedence field or DSCP Other IP and/or TCP information Combination of the above
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144
service scheduling and policing mechanisms next generation Internet: Intserv, RSVP, Diffserv, MPLS, 802.1p/q
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Reference
[1] ITU-T Recommendation G.729 Annex B: a silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications, Benyassine, A.; Shlomot, E.; Su, H.-Y.; Massaloux, D.; Lamblin, C.; Petit, J.-P.; Communications Magazine, IEEE Volume 35, Issue 9, Sept. 1997 Page(s):64 73 [2] Algorithmic and implementation aspects of echo cancellation in packet voice networks, Krishna, V.V.; Rayala, J.; Slade, B.; Signals, Systems and Computers, 2002. Conference Record of the Thirty-Sixth Asilomar Conference on Volume 2, 3-6 Nov. 2002 Page(s):1252 - 1257 vol.2 [3] Empirix, Inc., Hammer VoIP test system echo detection and analysis, 2001. [Online]. Available: http://wireless.feld.cvut.cz/mesaqin2002/Echo_Detection.pdf [4] GQ Maguire Jr., 2G1325/2G5564 Practical Voice Over IP (VoIP): SIP and related protocols, http://www.it.kth.se/courses/2G1325/VoIP-CoursepageSpring-2005.html, Spring 2005. S. Floyd and J. Kempf (Editors), IAB Concerns Regarding
Congestion Control for Voice Traffic in the Internet, IETF, RFC 3714, Network Working Group, March 2004. ftp://ftp.rfceditor.org/in-notes/rfc3714.txt
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