VoIP over P2P networksledvina/DHT/voip.pdf · –Interactive: User can control operation, e.g....

Víctor Ramos - UAM Iztapalapa October 7, 2006

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VoIPVoIP over P2P over P2P networksnetworks

VVííctorctor RamosRamos

UAMUAM--IztapalapaIztapalapa

RedesRedes y y TelecomunicacionesTelecomunicaciones

[email protected]

http://laryc.izt.uam.mx/~vramos

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• The IP protocol suite and related mechanisms and applications (what is standardized at IETF).

• An international network interconnection made of ISPs, enterprises, campus, etc.

• Philosophy:

– Intelligence at the edges.

– Fastest possible forwarding of packets.

– No states in the network.

What is the What is the ““InternetInternet”” ??


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Internet service: Best effortInternet service: Best effort

• Packets are routed throughout the Internet without any guarantee by IP (Internet Protocol):

– Packets can be lost: Congestion in routers, transmission errors over wireless links, link outage, etc.

– Packets can be delayed: Queuing in routers, handoff in wireless networks, retransmissions at the link layer, etc.

– Packets be replicated.

– And packets can be delivered out of order.

• TCP improves this service for data transfer applications ...


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Audio over the Internet: Audio over the Internet: AdvantagesAdvantages

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Audio over traditional Audio over traditional PSTNsPSTNs


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Convergence of voice and data Convergence of voice and data netsnets

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Business scenarios to support IP Business scenarios to support IP telephonytelephony


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Audio over a packet switching Audio over a packet switching network as the Internetnetwork as the Internet

• Audio signal is divided into time frames:

Frame Size

• Each frame is sampled, coded and sent in a separate packet.

• Look ahead: A codec may wait a little before coding a frame in order to get more information on the audio signal, which may improve the coding.

• Many codes exist, with different coding rates and different qualities.

• Silence detection and suppression: Most codecs detect the silence and stop the transmission until the speaker resumes talking.

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Rating quality with the ERating quality with the E--modelmodel• E-model: A model merging objective and subjective measures to predict the subjective quality a user will give to a voice call. The quality is evaluated using a subjective measure R that ranges from 0 to 100.

– The higher the rate, the better the quality.

– The quality of audio is acceptable if it is within the range 70-100 (comparable to the quality perceived in Public Switched Telephone Networks).


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Classes of audio applicationsClasses of audio applications

• Three classes of applications:

–Streaming

–Unidirectional Real-Time

– Interactive Real-Time

• Each class might be broadcast (multicast) or may simply be unicast (point-to-point).

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Audio streaming classAudio streaming class

Data stream

Playback pt Rcv pt

In transmission

or to be transmitted

• Streaming: Application usage of data during its transmission.

– Clients request audio files from servers and listen to them during the download, e.g. RealPlayer.

– Interactive: User can control operation, e.g. pause, resume, fast forward.

• Important and growing application due to reduction of storage costs, increase in high speed net access from homes, enhancements to caching and introduction of QoS in IP networks.

• No strict constraints on the delay (5 to 10 seconds judged acceptable).


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The other two classesThe other two classes

• Unidirectional Real-Time:

– Similar to existing radio stations, but delivery on the network.

– Non-interactive, just listen.

– No strict constraints on the delay, rate and losses are important.

• Interactive Real-Time :

– Phone conversation.

– More stringent delay requirement than Streaming and Unidirectional because of interactivity (good for less than 150 ms, acceptable for less than 400 ms, poor otherwise).

– This class is what we call IP telephony.

– Audio over the Internet or VoIP can be any one of the three classes.

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Transport of audio over the InternetTransport of audio over the Internet• These applications cannot be supported by TCP for different reasons:

– TCP introduces delay variability due to retransmissions and reordering.

– TCP introduces rate variability due to congestion control.

– Audio applications are sensitive to delay and rate.

• Moreover, audio applications can tolerate some losses and do not require full reliability:

• So the best choice is to support audio applications over UDP.

• But UDP does not provide any guarantee and the service can be very poor. Some measures must be taken !


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Internet service can be badInternet service can be bad

• UDP probes sent at regular time intervals

between INRIA and the University of

Maryland in July 1992.

– A probe every 500 ms.

– Round-trip time (RTT) is measured.

– Round-trip time set to 0 when a packet

is lost.

RT

T (

ms)

Probe number

9 % of probes are lost in this experiment.

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Challenges in supporting audio: Challenges in supporting audio: DelayDelay

• The delay can be large:

– Large delay harmful for interactive audio:

• One-way delay < 150 msec very good.• One-way delay < 400 msec acceptable, beyond 400 msecinteractivity is poor.

– And large delays exacerbate the problem of echo.

– Delay is not a problem for non-interactive audio e.g. radio, clips.

• Different components of delay:

– Propagation delay on links: NOTHING TO DO.

– Queuing delay in network routers: Something can be done.

• Give audio packets priority over data packets in network routers: Quality of Service.


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Solve jitter by delaying Solve jitter by delaying packetspackets

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FEC via variable encodingsFEC via variable encodings• Media-specific approach.

• Packet contents:

– High quality version of media frame k.

– Low quality version of media frame k-c (c is a constant called offset).

– If packet i containing high quality frame k is lost, then one can use packet i+c with low quality frame k in place.

1 2 3 4

2 3 41 2 31

Original audio stream

Addition of FEC (c = 1)

Transmission

3 42 31 Received stream (2 lost)

3 41 2 After decoding FEC

0

0


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Peer-to-peer Networks

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DefinitionDefinition ofof P2P*P2P*

1. Significant autonomy from central servers.

2. Exploits resources at the edges of theInternet– G storage and content

– G CPU cycles

– G human presence

3. Resources at edge have intermittentconnectivity, being added & removed

*Tutorial on P2P: Dan Rubenstein and Keith Ross, Infocom2004


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ApplicationsApplications deployeddeployed overoverP2P P2P networksnetworks

• P2P file sharing– Napster, Gnutella, KazaA, eDonkey, etc.

• P2P communication

– Instant messaging.

– VoIP: skype, netappel.

• P2P computation

– seti@home

• DHTs and their apps

– Chord, CAN, Pastry, Tapestry.

• P2P apps built overemerging overlays– PlanetLab

• Wireless ad-hocneworks.

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P2P file P2P file sharingsharing• Alice runs P2P clientapplication on hernotebook computer

• Intermittentlyconnects to Internet; gets new IP addressfor each connection

• Registers her content

• Asks for “Hey Jude”

• Application displaysother peers that havecopy of Hey Jude.

• Alice chooses one of thepeers, Bob.

• File is copied from Bob’s PC to Alice’s notebook: P2P.

• While Alice downloads,other usersuploading from Alice.


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• Is success due to massive number ofservers or simply because content is“free” ?

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SkypeSkype: P2P : P2P VoIPVoIP

• VoIP client developed in 2003 by theorganization that created Kazaa.

• It claims that it can (almost) cross NATs andfirewalls.

• Better voice quality than other VoIP clients(?)

• It encrypts calls e2e.

• It stores user information in a decentralizedway.

• It also supports instant messaging andconferencing.


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SkypeSkype architecturearchitecture

• Two types of nodes: Ordinary hosts andsuper nodes.

• Ordinary host (SC): Skyp app used toplace voice calls.

• Super node (SN): Ordinary host’s end-point on the Skype network.

• Any SC with enough resources is a candidate to become an SN.

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SkypeSkype architecturearchitecture (2)(2)• SC connects to an SN and authenticates itselfwith the Skype login server.

• The Skype login server is not a Skype node. Itstores usernames and passwords as well as buddy lists.

• There are SkypeOut and SkypeIn servers: they provide PC-PSTN and viceversa connectivity.

• Online and offline user information is storedand propagated in a decentralized way.


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SkypeSkype architecturearchitecture (3)(3)

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SkypeSkype architecturearchitecture (4)(4)

• Each SC maintains a table of reachablenodes called host cache (HC).

• An HC is stored in an XML file andcontains IP addresses and port numbersof super nodes.


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World World mapmap ofof supernodessupernodes

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SkypeSkype vsvs otherother VoIPVoIP clientsclients

• Why Skype users report better voicequality than with other VoIP clients?–Better playout delay control mechanism?

–Better loss concealment mechanism?

–Both?

– Is it due to the dynamics of P2P traffic?

• Maybe neither of these: Skype is a selfish application.


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SkypeSkype vsvs otherother VoIPVoIP clientsclients(2)(2)

• It tries to get the best available network andCPU resources during its execution.

• It changes its priority on Windows while a callis being established.

• Comparing the mouth-to-ear delay of YahooMessenger, Google Talk, and MSN Messenger with Skype, the latter reports the better m2e delay.

• So, under the same network conditions, it ismore likely that the best subjective voicequality is reported for Skype clients.

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BlockingBlocking SkypeSkype traffictraffic

• Why ?

• Corporate nodes are potentialcandidates to become SN.

• But this would relay unwanted trafficthrough the corporate network.

• But, blocking Skype traffic can becomevery difficult.


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HowHow toto blockblock SkypeSkype traffictraffic??• One must inspect the payload of the network (TCP, UDP) traffic.

• At login Skype sends a login message to the login server. The first two messages in that flow are:Skype LS

0x1603010000 -> (5 bytes)<- 0x1703010000 (5 bytes)

• By blocking all incoming messages who have the signature 0x17030100, Skype is blocked.

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HowHow toto preventprevent a a nodenode toto becomebecome anan SN?SN?

• One choice is to put a bandwidth limiteron Skype when no call is in progress.

• This is a quite “un-social” way of using Skype.


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ConclusionConclusion

• Skype differs significantly from otherP2P file sharing applications: mainly onthe user behavior.

• P2P VoIP shows as an attractive solutionfor real-time voice transmission over theInternet.

• Besides Skype, there is also Netappel, and others to come.

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ReferencesReferences• An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol.

Salman A. Baset and Henning G. Schulzrinne. Infocom 2006.• An Experimental Study of the Skype Peer-to-Peer VoIP System. Saikat

Guha, Neil Daswani and Ravi Jain. IPTPS '06.• Jean-Chrysostome Bolot, "Characterizing end-to-end packet delay and loss in the Internet", Journal of High Speed Networks, 2(3):305-323, 1993.

• Brian Gracely, “Voice over IP (VoIP)”, Cisco Systems Next Generation Internet Symposium, 2001.

• R. Ramjee, Jim Kurose, Don Towsley and Henning Schulzrinne, "Adaptive playout mechanisms for packetized audio applications in wide area networks," Proceedings of IEEE INFOCOMM, pp. 680--686, June 1994.

• Prof. Dan Rubenstein, “Real time Internet lecture”, Columbia University.• Prof. Henning Schulzrinne, “Tutorial on Voice over IP “, ACM SIGCOMM, august 2000.

• D. De Vleeschauwer, J. Janssen, G. H. Petit and F. Poppe, “Quality bounds for packetized voice transport”, Alcatel Telecommunications Review (1st quarter 2000) 19-24.

VoIP over P2P networksledvina/DHT/voip.pdf · –Interactive: User can control operation, e.g....

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