Download Voice-TFCC

Voice-TFCC: a TCP-Friendly Congestion Control
scheme for VoIP flows
Abdelbasset Trad
PRINCE Computer Science Research Unit
INFCOM Sousse, Tunisia
[email protected]
PIMRC 2008
15 Septembre 2008
Hossam Afifi
National Institute of
Telecommunications
INT Paris, France
[email protected]
Presentation Outline
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Motivations & Objectives
Studied Network Architecture
Congestion control for VoIP traffic
TCP-friendly Equation-based Rate Control
Our proposal: Voice-TFCC scheme
– Architecture
– Algorithm
– Analysis
• Conclusions and Perspectives
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Motivations of our Work
• VoIP is one of the fastest growing Internet applications
• The Internet is expected to carry a significant proportion of
the world’s telephony traffic
• New performance limitations have rised:
• Scale in number of VoIP communications
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Our Objectives
• Address the tradeoffs between efficiency and end-to-end
overall performance of a large number of VoIP
communications
• Design adaptive congestion control mechanisms aiming to:
• Maximize the overall VoIP transmission quality
• Use network resources efficiently
• Compete fairly with other Internet traffic (TCP)
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Studied Network Architecture
RTP Voice Flows
PSTN
Mobile Network
IP Network
VoIP
GW
PSTN
VoIP
GW
IP Phone
• Large number of VoIP sources at an access network destined to
different users in remote networks (PC to phone/Phone to Phone…)
• VoIP communications scharing a common path between peer VoIP
gateways
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Protocol Header Overhead
• Typical payload duration: 20 ms
Significant overhead
• IP/UDP/RTP encapsulation
• Minimum header length= 40 bytes
IPv6
Generated Header throughput:
16 kbps
IPv4
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Congestion Control for VoIP Traffic
• Voice traffic is typically deployed as best-effort traffic
• VoIP lacks effective and scalable congestion control
• Performance limitations:
• Inefficient use of network bandwidth (IP/UDP/RTP protocol headers)
• Fairness problem with TCP traffic caused by the transmission of large
number of uncontrolled UDP bursts of small VoIP packets
• UDP traffic is unresponsive to congestion and can completely
monopolize available bandwidth
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Protocol Header Overhead: Solutions
• Reduce the header size:
IP/UDP/RTP Header compression (cRTP, RFC 2508)
- designed for low speed serial links
- reduces IP/UDP/RTP header to 2 bytes (no UDP checksums)
- applied on a single RTP flow
- based on differential coding mechanism
- reliability ensured by lower layers
• Encapsulate several packets into one header (Multiplexing)
- multiplex several RTP streams between two gateways
- fixed number of flows to multiplex
- introduces delays (mutlipexing delay, queuing delays)
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TCP-friendly Equation-based Rate Control
• Unresponsive flows:
• Do not use end-to-end congestion control
• Do not reduce their load on the network when subjected to packet drops
• Multimedia applications using unresponsive transport protocols
(RTP/UDP)
• TCP-friendly equation-based rate control
• Introduced to ensure proper congestion avoidance for
multimedia applications
• Steady state TCP throughput approximation
• Smoothly find available bandwidth
• Do not halve the sending rate in response to a single loss
• Increase the sending rate slowly in response to a decrease of the
loss rate
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TFRC throughput as function of RTT and drop rate
S
T (Bps ) 
RTT
2p
3p
 RTO (3
) p (132 p2)
3
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TFRC Throughput
Payload size S=1460 bytes
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Our Proposal: Voice-TFCC Scheme
• Voice TCP-Friendly Congestion Control Scheme
• Novel generic scheme that controls both Packet and
Codec rate of VoIP flows while maintaining a TCPfriendly throughput
• Based on TCP-friendly decision:
– Packet rate is adjusted by multiplexing several RTP VoIP flows
over a single stream
– Codec rate is also adapted using different audio codecs
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Basic RTP Multiplexing Scheme
RTP header Payload
Source 1 IP UDP Voice 1
Source 2 IP UDP Voice 2
Source n IP UDP Voice n
…
IP UDP Voice 1 Voice 2 .. Voice n
Adjustable Aggregate Buffer
• Basic assumption: VoIP flows from different sources accur simultanousely at
the sender VoIP GW
• Number of packets to multiplex ?
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Voice-TFCC Architecture
• Transcoding module incorporated at the Voice-TFCC sender gateway:
- to handle homogeneous flows using the same voice codec rate
- to adapt this rate according to TCP-friendly decision
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Voice-TFCC Algorithm
Initially: m0=1
Case of one VoIP flow
Phase I: Reduce packet rate by multiplexing
Phase II: Reduce codec rate also
• How to determine the new packet and codec rate
to be used in the next time interval (i, i+1) ?
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Voice-TFCC Algorithm
Basic equation:
Phase I: Reduce packet rate by multiplexing
Phase II: Reduce codec rate
TFRC throughput formula
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Voice-TFCC Analysis: Bandwidth Saving
• Bandwidth saved by multiplexing m RTP voice packets:
where
Payload=20 bytes
Payload=160 bytes
• Multiplexed packet size is bounded by the MTU
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Conclusions & Perspectives
• Congestion control mechanisms for VoIP flows represents a promising
solution to prevent the performance degradation of voice and TCP traffic.
• Proposed Voice-TFCC mechanism dynamically adapts packet and
codec rate of VoIP flows while being fair with coexisting Internet traffic
A promising extention to Voice-TFCC scheme:
Study the case of high traffic load caused by a large number of flows that
can not be multiplexed within TCP-friendly flow
• Path switching mechanism: incoming VoIP flows are redirected towards a GW
presenting better network path conditions (signaling protocols like SIP can be
used)
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Thank you !
Questions ?
Dr. A. Trad
Voice-TFCC: Experimental Results
• Prototype implementation of VoIP traffic generation based on UDP sockets.
• An adaptive system that can switch between five bit-rates:
Different payload sizes
• PlanetLab hosts used to emulate VoIP GWs
• Feedback reports are sent from destination to sender host evry 5 seconds
• Initially 10 flows sent from host1 to host2 using G.711 codec
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Voice-TFCC: Experimental Results
P
• Introduction
Voice-TFCC
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MOS and Voice-TFCC/TCP-friendly rate difference
P
• Introduction
• TCP-friendliness condition achieved
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Experimental Results summary: Voice-TFCC mechanism
• Introduction
• Slight loss rate and jitter increase
• Significant delay and quality Improvement
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VoIP variant of TFRC (Floyd et al.)
• TFRC variant for applications that transmit small packets
• Assumption:
– Network bandwidth limitation in bytes/sec rather than in
packets/sec for VoIP traffic
• Design goal:
– Achieve the same bandwidth in bytes/sec as a TCP flow
using 1500 – byte data packets
• Penalize VoIP applications that send small payload packets
which increasing header overhead
– Reduce the sending rate (rate reduction factor)
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VoIP variant of TFRC
• The lower the payload size is the more
sending rate will be reduced
Voice packet size
TFRC throughput
Header size
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Packet-based vs. Byte-based Environments: Floyd’s Simulation Results
• Introduction
• 5 TCP connections and 5 VoIP TFRC connections sharing a 3 Mbps link
• In packet based environments each packet requires a single buffer and the
decision to drop a packet is independent of the packet size
• In byte-based environments small VoIP packets encounter less packet drops
than TCP
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Packet-based vs. Byte-based Environments: Floyd’s Simulation Results
• If the bottleneck link is in units of bytes rather than in packets:
• Fairness results change significantly
• VoIP TFRC flow sees a much smaller drop rate than TCP flow
• Consequently VoIP flow receives a much larger sending rate
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Experimental Results summary: Voice-TFCC mechanism
Phase I: without codec rate adaptation
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Introduction: VoIP Networks
• Initial interest of communication cost reduction (entreprise telephone
networks, long-distance calls)
• Now considered as networks that will replace the telephone network
• The base for the next generation of multimedia communications
• Flexibility of IP-based packet switched networks
• Convergence of data (packet switched) and Voice (traditionally circuit
switched) into a single IP-based core architecture.
• A single converged network for voice and data will be used
• VoIP services are being increasingly offered to end users (e.g. Skype)
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Overview: VoIP Quality Assessment
• ITU-T E-Model (G.107 Recommendation)
R 94.2  Ieeff  Id
Codec and loss Impairments Delay Impairment
• MOS (Mean Opinion Score)
1
for R0
MOS  10.035R7.106 R(R60)(100R) for 0R100
for R100
4.5
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Overview: VoIP Quality Assessment (2)
Effect of packet loss
Best Intrinsic codec quality
150 ms
Effect of delay
G.711
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Overview: VoIP Quality Improvements
• Research work focused on enhancing the low VoIP quality
related to intrinsic properties of IP networks
• Network QoS approach (DiffServ/IntServ)
• End-to-end mechanisms (Application level)
• FEC (Forward Error Correction)
• Playout buffer mechanisms (alleviate the jitter effect)
• Adaptive mechanisms
- End systems measure the service being delivered by the
network (using RTCP)
- Adapt their behavior according to packet delays and losses
- Adaptive FEC/Playout buffer mechanisms
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Overview: VoIP System
• Voice Codecs for analog voice digitization and compression
• Different techniques & different features
• Voice transport over best-effort IP networks
• RTP/RTCP over UDP
• No performance garantees
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Introduction: VoIP main Advantages
• Cost saving
- low Internet communication cost (packet switching technology)
- toll charges associated with PSTN networks are reduced
- reduced administration cost of a converged network
• Efficiency
- VoIP achieves more efficiency than the circuit-switched voice
transmission
- VoIP dramatically improves bandwidth efficiency
(advanced voice compression techniques, silence suppression)
• Integration of voice and data networks
- integrated networks intend to provide voice transmission quality
and reliability of PSTN networks
- combine voice communications with other media (e.g. video)33
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