29 - VoIP laboratory work: Signalling, Voice Quality and Security

Transcription

1 Helsinki University of Technology Networking Laboratory S Networking Technology, laboratory course 29 - VoIP laboratory work: Signalling, Voice Quality and Security Made by: Modified: Ilkka Kiiskinen ikiiskin@cc.hut.fi Pasi Lehto Supervisor: Vesa Kosonen Last update:

2 Introduction Voice over Internet Protocol, VoIP, has been on spotlight during the past few years. VoIP is a technology which allows you to make a telephone call over a data network, like Internet. This laboratory work gets you familiar with signalling, codecs, quality of service and security in VoIP-technology. QoS preliminary questions (total 46 points) PQ1: VoIP has been considered as a killer application by many specialists. Which economical and technical benefits can be achieved by implementing VoIP? What about the disbenefits? max. 7 points PQ2: Silence compression. What do the following acronyms stand for? Explain their functions briefly: VAD DTX CNG max. 3 points PQ3: Get familiar with the most typical audio codecs: G.711, G.723.1, G.729A and AMR. Find the main characteristics for those speech coders (sampling rate, bit rate, frame size (if framebased)). max. 6 points PQ4: Explain Absolute Category Rating (ACR) and Mean Opinion Score (MOS) value? How are they related with each other? What are the MOS values of G.711, G.723.1, G.729A, GSM and AMR codecs? max. 5 points PQ5: What does SPIT acronym stand for? Tell briefly (max. four sentences) about it. PQ6: Find the ITU-T G.114 recommendations for one-way transmission time. What are the reference values for the different one-way delay classes? max. 2.5 points PQ7: Ping one national, one Mid-European and one Australian address. Pick up the pinging values of each. You will use them in the laboratory work. Mention the addresses and the values in your preliminary report. max. 1,5 points

3 Signalling preliminary questions PQ1: Compare SIP (Session Initiation Protocol) and H.323-protocol. Compare at least six (6) features. max. 6 points PQ2: Mention the architectural components of SIP and H.323 protocol. Explain the function of each. max. 9 points PQ3: There are two types of gatekeeper signalling methods in H.323, which are they? Draw the signalling messages of a succeeded call signalling procedure in both methods. max. 4 points Read the working instructions and the questions, so that you have already a picture of the work in your mind. So you may find the answers for the final report questions a bit easier. As well you may skip prefaces during the work if you have already read them forehand and you may complete the work a bit quicker.

4 The assistant should make sure that the network is functioning (already at the beginning of the laboratory work) and Dummynet is already started, and the pipe is configured as follows: if Dummynet is off, turn it on give the login name (root) and the password (setup) type the following commands: sysctl -w net.link.ether.bridge=1 sysctl -w net.link.ether.bridge_ipfw=1 ipfw add pipe 1 all from any to any ipfw pipe 1 config bw 10Mbit/s Wait awhile, because the devices boot themselves. Check that the Siemens IP-phone is ok by calling to the other phone. VoIP laboratory work (total 53 points) The work is about VoIP signalling, QoS, codecs and VoIP security. The main stress is on the signalling and QoS parts. In the signalling part, you will focus on the H.323 and SIP protocol. Siemens HiNet follows the ITU-T s H.323 standard. To test SIP functionality, you will use OpenSER (SIP Express Router). It is a very lightweight implementation of SIP, but has the basic SIP functions as proxy server, registrar, redirect server, location server and presence. There are only SIP software clients available to be used in this work. The MSN Messenger version 4.6 will work as a SIP client. You will use Dummynet software to simulate different kinds of network scenarios. You will also use different kinds of software to simulate VoIP traffic and to monitor the traffic. Answer the questions given in the paper during the work. If the question is marked with asterisk (*), you may skip it and do it later, as a home assignment. Work environment The picture (3) represents the working environment.

5 Picture 1: working environment The Siemens IP-phones are there for testing H.323 protocol and QoS. Analyzer, ccm, PC5, rc300, sip1.voip.lab, sip2.voip.lab and Dummynet is used in testing SIP signalling, QoS and security. Software In this laboratory work you get a peek from some software concerning VoIP service, QoS analyzing, network analyzing and network manipulation. It is recommended that you check information about them from the internet and get a bit familiar with them forehand. MSN Messenger (v used in this laboratory work) is SIP-based communication software. It provides ways to chat online via text, voice or video conversation. It provides presence service and in the latest versions much more features. In this work we use it to voice conversation and instant messaging. Ascom QoS Analyzer is a QoS-testing tool. It uses fixed sized packets and simultaneous transmissions to analyze the network capabilities. It measures e.g. delay, jitter and packet losses by simulating different kinds of (simultaneous) calls. User may manipulate the parameters of the sent data. The software may be used to simulate different kinds of codecs. It can simulate up to 32 simultaneous calls. Ascom QoS Analyzer is java-based software. It provides a solution to measure QoS parameters by a round trip delay measurement. The measurement is done between the QoS Analyzer and an Echo Server. The measurement is made as a Round-Trip-Measurement. This means that the QoS

6 of the network is measured for both communication ways. It is not possible to measure only one communication direction, but this measurement method gives enough information about the overall quality of a link. The QoS Analyzer's indicated delay is calculated as the average of all measured delays from the single VoIP data packets (you may define the amount of the used packets yourself). Jitter is measured as the standard deviation over all round trip delay measured for single VoIP packets. And packet lost is calculated according to a jitter buffer setting. The interval for the "Jitter Buffer" is set to "Avg. Delay" +/- "Jitter Buffer Size", all packets arriving outside this delay interval are treated as lost. These features and methods won't give you accurate results, but you will get good approximations about the parameters. Ethereal is a popular network analyzer software. It is very flexible and it provides many features to monitor the traffic of the networks. Ethereal is used by network professionals around the world for troubleshooting, analysis, software and protocol development, and education. It runs on all popular computing platforms, including UNIX, Linux, and Windows. Agilent analyze software is also a network analyzer software. It is quite similar software to Ethereal. The purpose of the usage of this software is to get you familiar with different kinds of network analyzing software. Dummynet is a flexible tool originally designed for testing networking protocols, and since then (mis)used for bandwidth management. It simulates/enforces queue and bandwidth limitations, delays, packet losses, and multipath effects. It can be used on user's workstations, or on FreeBSD machines acting as routers or bridges.

7 Working instructions Signalling H.323 signalling H.323 is a standard approved by the International Telecommunication Union (ITU) in order to promote compatibility (components, protocols, procedures) of real-time point to point and multipoint audio, video, and data communication over packet-based networks, such as the Internet. It defines for example how endpoints negotiate shared audio or video encoding, how audio and video fragments are encapsulated and sent, how endpoints communicate with their gatekeepers and how endpoints communicate through a gateway. H.323 uses TCP and UDP for communication. Reliable transmission is needed for control signals and data traffic. Unreliable transport is used for audio and video streams, which are time-sensitive. Start the laboratory work by studying H.323 protocol. The Siemens IP phones run on the H.323 protocol. One is on the table in the laboratory room and the other is in the room G212. The assistant will open the door for you. Start the Agilent network analyzer in the analyzer computer. Switch off the Siemens IP phone in the laboratory room e.g. by unplugging the power cord. Start the measurement (Agilent analyzer software) and plug the phone back on. Stop the analyser and study the packets captured by the analyzer software. Q1: What is the protocol used during booting between the phone and the gatekeeper and which signalling messages are used in registration process? Make a H.323 call. Make a call between the Siemens IP phones (A->B). Let the callee (B) be the one to end the call. Keep using the Agilent analyzer and capture the messages from a successful call. Save the captured messages and send the saved file to your box or save it to your own disk, so you can study them afterwards. Q2: Draw the messages between the phones and the gatekeeper. Give also the protocol used to transmit the message and optionally the message request/response type (<protocol>:<message type>:<request/response type>) max. 6 points

8 *Q3: Explain how the call proceeds by following the signalling messages. max.6 points Q4: Which codec is used? Is the silence detection on? Where did you find the information? max. 3 points Q5: On which protocol the RTP packets are sent? max. 1 point SIP signalling Next you will study IETF s SIP (Session Initiation Protocol) protocol. The aim of SIP and some of its supporting protocols (e.g. SDP, ENUM and TRIP) is to offer a solution for global IP based voice and multimedia signalling and especially session establishment. OpenSER (SIP Express Router) is used in this exercise. There are two SIP proxies installed in the laboratory, sip1.voip.lab and sip2.voip.lab. To operate with the DNS domains they have the mentioned aliases. Another is responsible for the area sip1.voip.lab and the other for the area sip2.voip.lab. The users are registered by using URLs of type user@sip1.voip.lab or user@sip2.voip.lab. In this work you use Ethereal analyzer software on the computer rc3000 and the MSN Messenger version 4.6. The version 4.6 of the MSN Messenger still uses raw SIP protocol. For the latest versions MSN has added their own protocol MSNMS for authentication purposes, but it is still based on SIP. Use the Ethereal network analyzer software to capture SIP messages. Use the Ethereal in ccm. Start capturing traffic and make a normal phone call between Elvis and Elton (or Elvis and Bono - no matter). Q6: Draw a message sequence chart of a succeeded call setup and teardown between caller proxy 1 proxy 2 callee. Use help from literature. Mark in the picture which messages you got from the capture. max. 5 points *Q7: Take one INVITE message from the capture and explain its SIP header fields. max. 5 points Q8: MSN Messenger uses SIP presence service. It shows the states of the members in your contact list. Change the state of Elvis to "busy", "away" and back to online. Use Ethereal to capture all the signalling messages. How is the presence information sent between client and server? Q9: Make an IP call from Bono to Elton. Try then make a call from Elvis to Elton (busycase). What are the signalling messages in this case? Draw them in your report. *Q10: SIP has faced problems with NAT and firewalls. What are the problems? max. 3 points

9 QoS and codecs VoIP is built on the IP-networks, on the Internet. The Internet provides still best effort services. For traffic like VoIP it is vital that delay, jitter and packet loss stays small. So some QoS monitoring should be done for the VoIP-traffic. We should focus more on the end-to-end quality of VoIP, which is the factor of which the end user is interested in. The end user of VoIP will and should not be bothered by the technical terms, the technologies and the parameters underlying and defining the QoS of a connection. So the specialists that are working around the VoIP issues should pay attention to them more carefully. Codecs are used to convert an analogue voice signal to digitally encoded version. Codecs vary in sound quality, bandwidth required, computational requirements, etc. The codecs can be used to handle small bandwidth networks. Next we will study VoIP-traffic in different kinds of network circumstances. We use Dummynet software to simulate bandwidth limitations. We will use Ascom s QoS Analyzer software to simulate different VoIP traffics. We will also use the Siemens IP-phones to test and monitor the quality changes in the voice quality when packet loss ratio and delay alter. Dummynet will be used also to simulate different packet loss rations and delays. In the next assignment you will alter delay and packet loss values of the network. Use Dummynet to increase delay. Delay of the Dummynet is set by using the following command: ipfw pipe 1 config delay NN where NN is the delay value (e.g. 50ms) The packet loss ratio is set by using the following command: ipfw pipe 1 config plr X where X is a floating point number between 0 and 1, which causes packets to be dropped at random (e.g. 0.25). Q1: Make VoIP calls between the Siemens IP-phones and give grades (from 1 to 5 - both students give an individual grade) for the call quality in different scenarios. Ask quick questions and try to answer as soon as possible. You may also try to count from 1 to 10 as quick as possible, so that you count alternately. Use the values you got from the pinging task in the preliminary questions. Use also values 0ms, 500ms and 1000ms to simulate calls from a local call to a satellite call. If the phones get jammed, boot them. *Calculate the MOS values for each of the scenarios. Give and analyze the results with few sentences in your final report. Now alter the packet loss ratio and do the same grading again. Remember to change the delay back to zero (0). Q2: Use values 0, 0.05, 0.1, 0.16 and Close the call and make a new call setup after you have changed packet loss ratio. Try to make call at least three times if setup is not succeeded. Use long sentences. *Calculate the MOS values for each of the scenarios. Give and analyze the results with few sentences in your final report.

10 Set the packet loss parameter back to zero. Next you will get familiar with different kinds of codecs and the effects to the resource usage by changing the codec. Start the Echo Server in the computer PC5. Start the QoS Analyzer in the ccm. You can launch both of the programs from the start-menu. The QoS Analyzer can be also started from the command prompt by using following command (you should be in the QoS_Analyzer_ directory): java.exe jar QoS_Analyzer_ jar Create three new sessions - if they are not in the list yet, they should be there. The sessions should simulate G.711, G.723 and G.729 traffic. Instructions to setup the sessions: From the VoIP Stream -tab change the number of calls from 1 (Start) to 6 (End), increase the number of calls by one in each stream. Use Random as a simulation mode. Change the payload type (G.711, G.723 or G.729). To the Addressing -tab you should give the echo server address (PC5 IP-address). To the Session -tab, change the name of the session (e.g. G.729_1-6calls_12streams). No other changes should be made. When you have made the three different kinds of sessions, configure the Dummynet. In this part alter only the bandwidth. Use 1Mbit/s, 512Kbit/s and 256Kbit/s as the bandwidth values. The bandwidth with the Dummynet is changed by the following command: ipfw pipe 1 config bw NN Where the NN is the bandwidth (e.g. 256Kbit/s) Q3: Run each session in each different bandwidth scenarios. Monitor how well the different streams will manage in the different bandwidth limitations. Make a table to represent your results. Mark to your table how many simultaneous, acceptable calls can be made in the scenario in question - concentrate especially on packet loss and delay values. *Analyse the results. Do not concentrate too much to the values, but focus in comparing the codecs. max. 3 points

11 Security and threats VoIP hasn t survived without security problems. Companies implementing VoIP services tend to be concerned about quality issues and ignore security. The threads are mostly similar as for any of the data and the securing methods don t differ from the securing methods of the other data either. The differences concern mostly about architecture and some service features. This part scratches only the surface of VoIP-security. The idea of this part is to show how unsecured VoIP-traffic may be. Start the Ethereal in the ccm. Send instant messages between MSN Messenger clients Elvis and Elton. Capture the traffic during the instant messaging session. Q4: You may find the typed messages from the captured packets. Where? What information do you find from the end users? max. 3 points Keep using MSN Messenger. Make a call between Elton and Elvis. Use the head sets. Keep talking to each other about 20 seconds. Capture the traffic again. Filter the RTP traffic from the rest of the traffic. If voice data is seen only as UDP traffic, mark one of the RTP packets, choose Analyze/Decode as and choose RTP from the list. Now RTP traffic can be filtered from the rest of the traffic. Mark one of the RTP packets and choose "Statistics/RTP/Show All Streams ". Mark one of the streams (one with known payload (G.711)). Push the Analyze button. Save payload as.au file and listen the file with Windows Media Player. If you have only one headset, you should save the conversation only from one direction (simplex). Did you succeed? Let the assistant hear the captured conversation. *Q5: What security threats can be found from VoIP traffic or system? Mention at least four of them. (One of them may be closer than you thought) max. 4 points *Q6: Give few tips to improve VoIP security. max. 4 points Final report Answer the questions given in working instruction part and return your report via reservation system. It is recommended that you start writing the final report right after the working in the laboratory, so that the work is still fresh in your memory. The grading is composed from preliminary report, final report and working in the laboratory as follows: Max {PREL*0.66+FINAL+WORKING IN LAB} = 46* = 96 If you have any comments about the working instructions, please send them to ikiiskin@cc.hut.fi