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Open Source VoIP Traffic MonitoringLuca Derintop.orgEmail: [email protected]://luca.ntop.org/AbstractThese days voice over IP (VoIP) is quite popular as it is a cost effective ayto reduce telephony costs using the Internet. Although many projects arefocusing on developing tools and solutions for building the voiceinfrastructure, there is very little available in terms of tools and metrics formeasuring the impact of VoIP on a network.This paper describes the design and implementation of open source toolsfor detecting and measuring VoIP traffic based on both standard andproprietary protocols.KeywordsVoice Over IP, passive packet capture,network traffic measurement, NetFlow.1. IntroductionVoIP is a solid technology available since some years that allows people tocommunicate via voice using the IP protocol instead of telephone lines. Unfortunatelythis technology has been relegated in a niche market due to several factors such asproprietary standards, high price tag, limited integration with existing telephonyenvironments. However in the last couple of years the situation changed dramaticallysince some open source tools such as asterisk [asterisk] as well as low-cost VoIPtelephone adapters and services become available. In fact, today it is quite common forinternet providers to provide their customers VoIP calls at very low cost, if any, inaddition to standard xDSL connectivity.Standard VoIP protocol such as SIP [sip] and H.323 [h323] are very popular in thecarrier environment and in many other fields not limited to VoIP, such as messenger andchat. In addition to these standards-based applications, there are other applications suchas Skype [skype] or voipstunt [voipstunt] that instead are based on proprietarycommunication protocols and codecs, and other hybrid applications partially based onopen standards such as google talk [googletalk] and gizmo [gizmo]. The result is thatVoIP is becoming in some ways similar to P2P (peer to peer), as: new applications appear, grow and disappear very often. some VoIP applications (e.g. Skype) are using P2P as communication transport forbuilding the communication infrastructure and crossing firewalls, a typical scenariowhere many standard-based VoIP application fail to operate.In a nutshell, VoIP solutions are often used at corporate level as a cost effective solutionto telephone communications, whereas proprietary VoIP applications are used for lettingpeople talk either computer-to-computer or computer-to-telephone using a PC equippedwith a special application and a headset.

2. MotivationIn this complex and evolving scenario, VoIP traffic monitoring tools are very few, oftenintegrated into packet sniffers such as ethereal [ethereal] [hollis] and used for findingissues (e.g. severe packet loss or incompatible codecs) in specific situations, rather thanfor permanently monitoring VoIP and non-VoIP traffic. Other tools such as Vomit[vomit] or RTP-tools [rtp-tools] are suitable for capturing voice communications but notfor providing a comprehensive permanent monitoring tool. This has been the authormotivation for this work, namely to develop an open source VoIP-aware trafficmonitoring tool able to: Provide long-term monitoring, contrary to what available VoIP monitoring tools do. Handling standard VoIP protocols as well, as much as possible, proprietary protocols. Decode calls, hence identify peers (who’s calling who) and client applications. This isuseful for VoIP accounting, billing or fraud detection. Provide VoIP metrics such as packet loss and latency, as well as voice quality. Generate traffic trends in order to identify how VoIP traffic is changing over the time.In order to achieve the above goal, the author decided to use a dual approach: Enrich ntop [ntop], a home-grown open-source passive traffic monitoring application,for making it VoIP traffic aware. Develop some metrics suitable for monitoring key VoIP traffic characteristics andexport them via Netflow [netflow] v9/IPFIX [ipfix], by means of nProbe [nprobe] anopen-source netflow probe also developed by the author.This decision has been made because: It allows users to exploit the available traffic analysis facilities provided by ntop,without having to run any specialized VoIP traffic analysis application. In this wayVoIP traffic is not treated as first-class citizen but it is at the same level as other traffic(e.g. http or email). It enables VoIP measurements computed by nProbe to be exported using the standardNetflow/IPFIX protocol, so that they can also be used by ntop and other commercialnetflow applications such as Cisco NetFlow Collector. This is particularly importantwhen open source solutions are deployed in an enterprise that is using an existing/commercial management consoleThe following sections describes the design and the implementation of the extensions tontop and nProbe for monitoring VoIP traffic.3. VoIP BasicsAs stated before there are three main VoIP protocol families, namely those based on: standards protocols such as SIP/H.323/RTP [rtp]; proprietary but well documented protocols such as Cisco skinny [skinny]; proprietary protocols such as Skype.Note that the use of standard or known protocols does not always means that it ispossible to monitor everything as protocols such as RTP, used to carry voice and video,may transport data encoded with proprietary codecs. This is true for instance for GoogleTalk whose voice is encoded with a proprietary codec. In general all the protocols are

based on a connectionless protocol such as UDP.CallerCalledPartySignaling ProtocolVoice/Video DataFigure 1. - VoIP Protocol ArchitectureThe figure above shows the basic of VoIP. Every communication is made of three basicsteps:1. When a caller wants to communicate with another party it initiates a communicationeither with the remote party or with a gateway/PBX (this depends on the protocolbeing used and on the local network setup) using a signaling protocol. This step isresponsible for: Verifying party credentials, credit (if applicable), ability to call the specifiednumber; Negotiating the call (e.g. is it voice only or voice and video). Agreeing on a common codec (e.g. H.264). Negotiating the ports used for exchanging voice/video data.2. The call takes place on the ports previously selected, and the payload is encodedusing the specified codec. If a standard protocol is used, usually RTP is the oneselected. In case of a video-call there are two independent RTP streams, one forvoice and one per video.3. When one of the parties decides to complete the call, using the signaling protocolthe call is terminated.The following table lists some popular signaling and transport protocols used for VoIP.SignalingTransport SIP (Session Initiation Protocol) RTP (Realtime Transport Protocol) Cisco Skinny RTCP XS (RTP Control ExtendedReports) H.323Figure 2. - Popular Signaling and Transport ProtocolsFrom the traffic monitoring point of view: The signaling protocol contains important information such as parties identity, type ofcall (voice or video-call), codecs, duration, and information about the RTP session(s)that usually do not take place on fixed ports. In general without properly decoding thesignaling protocol, it is not possible to guess the ports used for RTP.

The voice/video transport protocol is used to extract information such as jitter, packetloss, and packet latency that are the building blocks for evaluating the call quality.Note that as the RTP packets contains information about the codec being used, withappropriate software it is also possible to decode the RTP and extract further callinformation. Protocols such as RTCP XS [rfc3611] are used to report information about RTPstreams including informations such as packet loss, burst/delay, call and transmissionquality metrics. These reports are not always available for VoIP calls as they are oftengenerated by VoIP equipments.For other closed and proprietary protocols such as Skype, it is not really possible todecode the call. Although some researchers tried to reverse-engineer the protocol[baset], to date it is not possible to grab any call information from the traffic stream.Actually the first problem is the detection of the phone call, as Skype is based on theeDonkey P2P [edonkey] protocol. Therefore at best a monitoring application can onlydetect the phone call and the parties IP, but not the party identities. For this reason, thiswork will focus only on (partially) open VoIP protocols as they allow monitoringapplications to decode relevant call information.4. Monitoring VoIP TrafficVoIP traffic monitoring is divided in two big families: proprietary and standard VoIPprotocol monitoring. This section describes how VoIP traffic has been monitored usingtwo open source applications developed by the author.4.1. Proprietary VoIP Traffic MonitoringThe main VoIP protocol that falls into this category is Skype. As the protocolimplementation is currently unknown and the packet payload encrypted, the best way tomonitor this protocol is to threat it as special eDonkey protocol communication.Typically Skype is detected as follows: The underlying protocol must be eDonkey. This can be detected by dissecting theinitial session payload as described in [karagiannis], and partially relying on thedefault port being used. Patterns searching for Skype detection has been implementedusing the popular PCRE [pcre] library. This library that allows patterns to beefficiently searched into into a data buffer, has been used to search for Skype patterninto the packet payload. The protocol pattern definition has been borrowed by thepopular l7-filter [l7-filter] tool that includes several patters not limited only to P2P/VoIP protocols. Thanks to this solution, it is possible to detect not only Skype ingeneral, but also the conversation type (skype2skype or skype-in/out call). As Skype traffic looks similar to the original eDonkey traffic, it is necessary to furthercharacterize the traffic in order to distinguish eDonkey from Skype. As protocolpayload is encrypted, the only choice left is the analysis of traffic conversations. Inparticular the main differences between a P2P and Skype conversation are: During a Skype conversation, traffic is bidirectional, packet frequency is high (ingeneral around 64 packets/sec regardless of peers speaking or not) with limitedjitter, packet size is limited (usually below 250 bytes). A eDonkey P2P session instead is mostly unidirectional (from the source of data tothe host where data is directed), packet rate is not constant and packet size is much

larger.In a nutshell the only thing that a monitoring application can do with respect to Skypetraffic, is to provide evidence of calls without furnishing any other information such thenickname of the people who held the conversation. For this reason Skype detection hasbeen implemented only inside ntop and not on nProbe as there are almost no metrics toexport while analyzing Skype traffic.4.2. Standard VoIP Traffic MonitoringAs explained before, VoIP traffic analysis is divided in two parts: Signaling protocol analysis. Voice traffic analysis.The author decided not to analyze legacy signaling protocols such as H.323 but insteadfocus on modern protocols or industry standards as SIP and Cisco Skinny. For voicetraffic analysis the choice is simple as RTP is basically the only protocol being used;this is because RTP has been designed flexible enough to carry various type of data (e.g.not only voice) coded in various formats.The implementation of VoIP monitoring is slightly different in ntop and nProbe. From auser survey, ntop users are more interested in having a “simple to use and understand”traffic analysis overview. Instead, nProbe users are usually professional networkadministrators, who prefer precise traffic metrics that can be meaningless for nonprofessionals. For this reason, ntop has been designed to provide VoIP traffic evidencewith some simple metrics, whereas nProbe sports precise VoIP traffic metrics that canbe used by netflow collectors for building accurate analysis applications. However it isworth to note that as ntop can act as a flow collector, ntop can also receive and takeadvantage of nProbe traffic metrics.The following table shows the metrics that nProbe is currently able to measure.SIP MetricsRTP Metrics SIP CALL ID RTP FIRST SSRC SIP CALLING PARTY RTP FIRST TS SIP CALLED PARTY RTP LAST SSRC SIP RTP CODECS RTP LAST TS SIP INVITE TIME RTP IN JITTER SIP TRYING TIME RTP OUT JITTER SIP RINGING TIME RTP IN PKT LOST SIP OK TIME RTP OUT PKT LOST SIP ACK TIME RTP OUT PAYLOAD TYPE SIP RTP SRC PORT RTP IN MAX DELTA SIP RTP DST PORT RTP OUT MAX DELTATable 1. - VoIP Traffic Metrics

Note that these metrics can be exported only using NetFlow v9 or IPFIX - supported byboth nProbe and IPFIX - as previous NetFlow version such as v5 have no room forcarrying extra information. Instead v9/IPFIX have the ability to define flow templatesthat dynamically define the flow format and attributes. Those metrics have beenimplemented in order to satisfy basic traffic measurements such as: SIP Unique call identifier used for accounting/billing and tracking problems. Call parties: caller and called party. Codecs being used, useful for identifying voice quality issues due to the use ofcodecs with poor quality. Time of important call events such as beginning of the call. These times can beused to identify performance issues on the SIP gateway. RTP ports where the call will take place. This information is necessary forassociating a signaling flow with the phone call just negotiated. RTP Source identifiers and time-stamp for the first and last RTP flow packet. Jitter calculated in both (in to out, and out to in) directions. Number of packets lost as well as maximum packet time delta in both directions. Identifier of RTP payload type as specified in [rfc2862].With the above metrics it is possible to create a wide range of measurement applicationssuch as simple “who’s talking to who” CDR (Call Data Record) used for accounting andbilling, and complex traffic analysis applications able to identify communicationproblems due to the use of poor codecs or high network jitter.nProbe implements VoIP support in two plugins, one for SIP and the other for RTP.VoIP measurements are exported inside v9/IPFIX flows using custom flow templates.The following example defines a simple SIP flow template whose identifier is 257.nprobe -n 192.168.0.1:2055 -U 257 -T "%LAST SWITCHED %FIRST SWITCHED %IN BYTES%IN PKTS %OUT BYTES %OUT PKTS %SIP CALL ID%SIP CALLING PARTY %SIP CALLED PARTY%SIP RTP CODECS %SIP RTP SRC PORT %SIP RTP DST PORT"Figure 3. - Simple NetFlow v9 flow template definitionAs stated before ntop is able to collect those flows and understand the VoIP metrics thatcan be defined into the flows. However as v9/IPFIX is an open architecture where flowformat is defined using standard templates, commercial applications such as CiscoNetFlow Collector are also able to collect flows and use the VoIP traffic measurements.NetFlow FlowsnProbentophttp(s)Network TrafficWeb BrowserFigure 4. - nProbe flow export towards ntop

The figure above depicts a common setup where ntop collects flows (including VoIPcalls) emitted by nProbe in NetFlow v9 format. Nevertheless ntop can also analyzeVoIP traffic without having to use flows as feeds. In fact it is also possible for ntop toanalyze traffic natively by means of the libpcap [pcap] library. From the user point ofview, there is virtually no difference from analyzing VoIP traffic with ntop using netflowor libpcap. This is because one of the main design goals of ntop is to hide differences tothe user in terms of traffic capture techniques or network interface types.Figure 5. - ntop: VoIP Session DetailFor each host that generates VoIP traffic, ntop puts an icon next to it. Clicking on thehost, ntop displays further information such as user alias or telephone number as seen inthe VoIP traffic.Figure 6. - ntop: Host DetailIn the sessions list, ntop lists the ongoing phone calls complete with call informationsuch as peer telephone number. ntop can also keep track of video calls as shown infigure 5, where two simultaneous sessions (one for voice and one for video) are activefrom the same peers. Basically ntop handles VoIP calls as sessions even if they arebased on UDP and not TCP, and reports call details into each session. As of today, ntopcan handle both Skype/Skinny/SIP/RTP, whereas nProbe handles only SIP/RTP.In case of Skype traffic, ntop puts an icon next to the host but as explained before it isnot able to display any additional call information. From users experience, the use ofpatterns for detecting skype traffic is quite reliable but not the ultimate solution. This isbecause sometimes the pattern is reporting false positives (e.g. non-Skype traffic issometimes marked as such) even on HTTP connections, definitively used for tunnelingSkype traffic. On the other hand the use of patters do not seem to have false negatives(e.g. inability to detect Skype traffic).Both ntop and nProbe have been deployed on networks with both proprietary andstandard VoIP traffic. The performance of the tools seems to be acceptable and their usehelped significantly to unhide details of VoIP communications. The main issue isinstead the way these tools are deployed, in case they need to be used for analyzing onlyVoIP traffic. In fact VoIP traffic is very limited compared to the overall traffic, in termsof both packets and bytes volume. This means that if a Gbit link needs to be analyzed,most of the work of these tools is packet discard of non-VoIP packets; this activity thatcan take quite some time and waste all the CPU cycles if some packet accelerationfacilities [deri] are not used. Furthermore as RTP traffic is flowing on dynamic ports,packet filtering facilities provided by standard equipment such as Juniper routers are notsuitable as they are static and not able to be reconfigured on the fly based on the

signaling protocol. The conclusion is that on fast links, it is advisable to either analyzeonly the signaling protocol without taking into account RTP, or use packet filtering andacceleration if RTP needs also to be used.5. Open Issues and Future WorkThe main open issue is the inability to properly handle Skype, to date probably the mostwidespread VoIP protocol. As explained before, this is due to the lack of documentationabout the protocol, and the use of payload encryption. This is not only a limitation ofntop and nProbe but of any other VoIP analysis tool.VoIP support is relatively new into ntop/nProbe hence several extensions can be addedto their implementation. The current measurements focus mainly on high-level metricssuch as jitter or packet loss, and are independent of the codecs being used. However asnew codecs such as H.264 [h264] are becoming increasingly popular, a plannedenhancement is the ability to decode some of these codec formats in order to alsoprovide precise information about the RTP payload (e.g. voice quality), as well providesupport for RTP XS reports. The implementation of these voice analysis metrics hasbeen delayed with respect to the original plan, as they are described in ITU documents(e.g. ITU E.411 recommendation) that are not freely available on the Internet, that isusually a problem for the open source community. Nevertheless in the next release twonew common metrics such as MOS score and r-factor will be implemented thanks tobits and pieces found googling on the Internet.6. Final RemarksThis paper described the challenges of VoIP traffic monitoring and presented two opensource traffic monitoring applications able to also monitor VoIP traffic.This work is novel in many aspects: Beside traffic sniffers, this is the first open-source traffic monitoring application ableto continuously monitor VoIP traffic. nProbe is probably the most flexible and advanced NetFlow probe available, anddefinitively the first probe able to monitor VoIP traffic using v9/IPFIX.Furthermore thanks to packet capture acceleration [ncap], it is also possible to monitorVoIP traffic on gigabit links using nProbe/ntop with almost no packet loss.7. AvailabilityThis work is distributed under the GPL2 license and is available at the ntop home page(http://www.ntop.org/) and other mirrors on the Internet (e.g. http://sourceforge.net/projects/ntop/).8. AcknowledgmentThe author would like to thank Urmet TLC (http://www.urmet.it/) for partially fundingthis research work.

9. References[asterisk]B. Schwarz, Asterisk open-source PBX system, Linux Journal, February 2004.[baset]S. Baset and H. Schulzrinne, An Analysis of the Skype Peer-to-Peer Internet TelephonyProtocol, http://arxiv.org/pdf/cs.NI/0412017, Columbia University, September 2004.[deri]L. Deri, Improving Passive Packet Capture: Beyond Device Polling, Proceedings ofSANE 2004, 2004.[edonkey]K. Tutschku, A Measurement-based Traffic Profile of the eDonkey Filesharing Service,PAM 2004[ethereal]G. Combs, Ethereal, http://www.ethereal.com/.[gizmo]The Gizmo Project, http://www.gizmoproject.com/.[googletalk]Google Inc., Google Talk, http://talk.google.com/, 2005.[h264]The MPEG Expert Group, MPEG-4 Part 10, ITU-T H.264,http://ftp3.itu.ch/av-arch/jvt-site/, 2005.[h323]H. Liu and others, Voice over IP Signaling: H. 323 and Beyond, IEEE Communications Magazine, 2000.[hollis]E. Hollis, Monitoring & Analyzing VoIP Traffic, Certification Magazine, February2005.[ipfix]B. Claise and others, IPFIX Protocol Specification, Internet Draft, 2005.[karagiannis]T. Karagiannis and others, Transport Layer Identification of P2P Traffic, InternetMeasurement Conference, 2004.[l7-filter]Application Layer Packet Classifier for Linux, http://l7-filter.sourceforge.net/.[ncap]L. Deri, nCap: Wire-speed Packet Capture and Transmission, E2EMON, May 2005.[netflow]B. Claise, NetFlow Services Export Version 9, RFC 3954, October 2004.[nprobe]L. Deri, nProbe: an Open Source NetFlow Probe for Gigabit Networks, TNC 2003,May 2003.[ntop]L. Deri and others, Monitoring networks using ntop, IM 2001, May 2001.[pcap]Lawrence Berkeley National Labs, libpcap, Network Research Group,http://www.tcpdump.org/.[pcre]P. Hazel, PCRE - Perl Compatible Regular Expressions, http://www.pcre.org/.[rfc2862]M. Civanlar and G.Cash, RTP Payload Format for Real-Time Pointers, RFC 2862,June 2000.[rfc3611]T. Friedman, and others, RTP Control Protocol Extended Reports (RTCP XR), RFC3611, November[rtp]H. Schulzrinne and others, RTP: A Transport Protocol for Real-Time Applications,RFC 1889, January 1996.[rtp-tools]H. Schulzrinne, RTP Tools, [sip]M. Handley, SIP: Session Initiation Protocol, RFC 2543,March 1999.[skinny]Selsius Inc., Skinny Client Control Protocol.[skype]D. Bergström, An analysis of Skype VoIP application for use in a corporate environment, http://www.geocities.com/bergstromdennis/, October 2004.[voipstunt]VoIPstunt, http://www.voipstunt.com/.[vomit]N. Provos, Vomit - voice over misconfigured internet telephones,http://vomit.xtdnet.nl/.

VoIP traffic monitoring is divided in two big families: proprietary and standard VoIP protocol monitoring. This section describes how VoIP traffic has been monitored using two open source applications developed by the author. 4.1. Proprietary VoIP Traffic Monitoring The m