Projekt (0)

Tallinna Tehnikaülikool Raadio- ja sidetehnika instituut
Projekt ainetes ,,IRT0030 Andmeside"
ja
,,IRT0100 Kommunikatsioonivõrkude struktuurid ja teenused"
teemal
« VoIP teenus»
Üliõpilane: Ruslan Karpovits Õpperühm: IATM Matrikli nr: 050829 Õppejõud: Avo Ots
Tallinn
2008 Author 's word This project is written to show some interesting aspects of working with VoIP ( Voice over Internet Protocol ) service. The project briefly describes the process of finding a solution for based VoIP problem and its realization.
-2- Contents
Introduction .............................................................................................................................. - 4 -
Mission ....................................................................................................................................... - 5 -
Solution...................................................................................................................................... - 5 - Scheme .................................................................................................................................... - 5 - Components ............................................................................................................................ - 5 - Configuration ......................................................................................................................... - 7 - Monitoring.............................................................................................................................. - 8 - Price ..................................................................................................................................... - 10 -
Conclusion ............................................................................................................................... - 10 -
Referencies .............................................................................................................................. - 11 -
-3- Introduction VoIP is a solid technology available since some years that allows people to communicate via voice using the IP protocol instead of telephone lines. Unfortunately this technology has been relegated in a niche market due to several factors such as proprietary standards, high price tag, limited integration with existing telephony environments. However in the last couple of years the situation changed dramatically since some open source tools such as asterisk as well as low- cost VoIP telephone adapters and services become available. In fact , today it is quite common for internet providers to provide their customers VoIP calls at very low cost, if any, in addition to standard xDSL connectivity.
The VoIP uses Internet protocol (IP) to send digitized voice traffic over the Internet or private networks . An IP packet consists of a train of digits containing a control header and a data payload. The header provides network navigation information for the packet, and the payload contains the compressed voice data.
While circuit -switched telephony deals with the entire message , VoIP-based data transmission is packet-based, so that chunks of data are packetized (separated into units for transmission), compressed, and sent across the network--and eventually re-assembled at the designated receiving end. The key point is that there is no need for a dedicated link between transmitter and receiver.
Packetizing voice data involves adding header and trailer information to the data blocks. Packetization overhead (additional time and data introduced by this process) must be reduced to minimize added latencies (time delays through the system). Therefore , the process must achieve a balance between minimizing transmission delay and using network bandwidth most efficiently --smaller size allows packets to be sent more often, while larger packets take longer to compose. On the other hand , larger packets amortize the header and trailer information across a bigger chunk of voice data, so they use network bandwidth more efficiently than do smaller packets.
Figure 1.Signaling and transport flows between endpoints.
-4- Mission We are a company that provides Voice over IP telephony over Estonia. Our mission is to make for some company IP telephony connection , which offices are located in Tallinn and Narva.
Figure 2.Range between customer offices
We need to provide them with fair speech quality (3,5 phones , as workers will be speaking through computer IP telephone applications. It is hoped that this will reduce some costs . So, we need to find a solution, whitch requires IP telephone connection itself, its monitoring and proper equipment .
Solution Scheme First of all we have made a simple scheme, which you can see on Figure 3.
Figure 3.The main scheme
Components Secondly, for our project we need proper equipment.
As a servers we use the computers with operating system Linux and an open source telephony engine Asterisk IP-PBX. Asterisk possesses all possibilities of classical automatic telephone exchange , supports most of VoIP protocols and provides functions of voice mail, conferences, -5- the interactive voice menu (IVR), the centre of processing of calls - statement of calls to turn and their distribution on agents using various algorithms, record CDR and other functions.
On the one hand, the server is connected with telephone lines and can incorporate to any phone of the world. On the other hand, the server is connected with the Internet and can contact any computer in the world. The server accepts a standard telephone signal , digitizes it (if it initially not digital ), considerably compresses, breaks into packets and sends through the Internet to destination point with use of the protocol TCP/IP. For the packets coming from the Network on a telephone server and going out in a telephone line, operation goes upside-down. Both making operations (a signal input in a telephone system and its exit from a telephone system) occur practically simultaneously that allows to provide fullduplex conversation .
In each office building we put a broadband router and the necessary quantity of switches. In our case we have 95 worksites, so we need 2 switches with 48 ports each.
As VoIP client we use Zoiper, which is a IAX and SIP softphone for Windows , Mac OS X and Linux. We chose it as its user -friendliness, diversity and richness of features set new standards for functionality and perfection. Also Zoiper offers great simplicity with enhanced interface to bring a smooth and satisfying VoIP communications experience.
Figure 4.The Zoiper' interface
Also it is necessary to choose the right audio codec for VoIP transmission which ensure us the right quality of MOS from 3,5 to 4. In this case the best choice for us will be a G.729 codec ­ a speech codec, with the low- latency G.729 audio data-compression algorithm, which partitions speech into 10-ms frames . It uses an algorithm called conjugate-structure ACELP (CS-ACELP). G.729 compresses 16-bit signals sampled at 8 kHz via 10-ms frames into a standard bit rate of 8 kbps, but it also supports 6.4-kbps and 11.8-kbps rates. In addition, it supports voice-activity detection and comfort-noise generation.
Video conferences are too important for our customer. For video codec we chose the default H.261. This standard, developed in 1990, was the first widely used video codec. It introduced the idea of segmenting a frame into 16 3 16 "macroblocks" that are tracked between frames to establish motion -compensation vectors. It is mainly targeted at videoconferencing applications over ISDN lines (p 3 64 kbps, where p ranges from 1 to 30). Input frames are typically CIF (352 3 288) at 30 frames-per-second (fps), and output compressed frames occupy 64 kbps to 128 kbps for 10-fps resolution.
-6- Configuration We need to configure our equipment for work . The most important part here is the configuration of Asterisk PBX stations.
Firstly, it must be mentioned that for our mission we use a SIP protocol (Session Initiation Protocol), which is used for creation , change and rupture of sessions: transfer of any data from the sender to the addressee between one or several participants. SIP describes only rules of connection between devices, therefore together with SIP the information transfer protocol is used. SIP uses for data transmission protocols RTP ( Real -Time Protocol) and SDP (Session Description Protocol).
So, let's make a simple configuration between two PBXs.
Figure 5.The scheme of SIP connection
Stations' data:
PBX 1 address: http://192.168.252.41 PBX 2 address: http://192.168.252.42 User: admin User: admin Pass : trikstriks Pass: trikstriks Phone Numbers : 100-199 Phone Numbers: 200-299
To configure a SIP connection between PBX 1and PBX 2, first of all we need to go to their addresses ( http://192.168.252.41 http://192.168.252.42 ). We write given User name and Password. Firstly, let configure SIP Trunks. From the menu of FreePBX => Add Trunk => Add SIP Trunk , where in Outgoing Settings we need to write the following : [Trunk name] ­ the name of trunk host ­ IP address or domain name of the peer PBX we want to connect to username ­ this is the name of the SIP trunk coming from incoming PBX secret ­ this is the one password that is used to connect BOTH SIP trunks together. We must use the same password at both PBXs qualify=yes ­ forces the trunk to register context =from-trunk - calls come in through the trunk
As a result : For PBX 1: For PBX 2: [200] [100]
host=192.168.252.42 host=192.168.252.41 username=100-user username=200-user secret= 1234 secret=1234 qualify=yes qualify=yes context=from-trunk context=from-trunk
In Incoming Settings we need to erase everything as nothing is needed in this section . So, the configuration of trunks is completed.
Now let's go to the next step - Outbound Routes. From the FreePBX menu => Outbound Routes, where it is needed to write the following:
-7- Route Name ­ must be a descriptive name of route Dial Patterns ­ example of extension number Trunk Sequence ­ this is the trunk that we configured
As a result: For PBX 1: For PBX 2: [200-dial2xx] [100-dial1xx] [2xx] [1xx] [IAX2/200-peer] [IAX2/100-peer]
The last step is to set some users or Extensions. All in all we need 25 users for PBX1 and 25 for PBX2. We will describe the creation of 2 users as the configuration of rest is absolutely the same. From FreePBX menu => Extensions where from Add an Extension menu we select «Generic SIP Device ». On page which comes after that the following parameters were required: User Extension ­ user phone numbers (as example, for PXB1 the number will be , for PBX2 - ) Display Name ­ user names (as example, for PXB1 user's name will be HOMER , for PBX2 ­ BART )
Device Options Secret ­ password
Voicemail & Directory Options Status enable ­ turn on the status Voicemail password ­ password of voicemail
By pressing the orange button " Apply Configuration Changes " and chosing " Continue with Reload", we finished the configuration of our SIP connection between stations PBX1 (192.168.252.41) and PBX2 (192.168.252.42).
Next step is the configuration of special OS application for VoIP Zoiper on differnet computers. The configuration of that is very simple. There we go to Sip accaunts => Add new SIP account , where we write the following settings: User on 1st PC: User on 2nd PC: Domain: 192.168.252.41 Domain: 192.168.252.42 (address of user's station ) Username: 150 Username: 250 (phone number) Password: 1234 Password: 1234 (personal password) Caller ID Name: Homer Caller ID Name: Bart (name for Zoiper)
As a result, now we can make telephone calls over IP between two users (Homer and Bart) from one PBX to another . To see, how our connection works, by using special application Wireshark we saved all packets during one IP telephoning session between Bart and Homer. By analyzing these packets we may say that SIP connection is working in this way:
The first user Homer dials number, the SIP-client generates signal INVITE ( invitation ), at the second user's (Bart) phone rings , its SIP-client gives out the message 180 (Ringing, a call ), then the user takes the call, the SIP-client gives out the message 200 (OK), the first SIP-client sends the second signal ACK (acknowledgement) and further is going the transfer of a vocal stream under protocol RTP (Real-time Transport Protocol) begins . When conversation is ended also one of users hangs up a receiver, the SIP-client sends signal BYE.
Monitoring Forthermore, we need to monitor or control our-made connection. For this purpose we use an application from Malden Electronics - VoIP Monitor Professional. This is a software tool for non-intrusive assessment of speech quality and measurement of RTP/RTCP traffic parameters in VoIP networks. VoIP Monitor analyses the RTP traffic to produce real-time useful statistical -8- data and predictions of speech quality across a single call or a number of calls. A calibration file, tailored to the gateway or VoIP terminal refines the speech quality prediction and gives a better correlated Mean Opinion Score (MOS) value than any other monitoring method .
Figure 6.The interface of VoIP Monitor Professional application
VoIP Monitor Professional analyses the Voice over IP traffic by monitoring live RTP/RTCP traffic streams. The performance of a VoIP transmission network can be characterised mainly in terms of Packet Loss and Jitter.
Figure 7.Jitter buffering and packet loss concealment
Packet loss can degrade speech quality but effective packet loss concealment can cover up these degradations. Jitter can also affect speech quality if it exceeds the capacity of the jitter buffer or
-9- the dynamic jitter buffer algorithm is inadequate. The jitter buffers, codecs and packet loss concealment techniques influence the speech quality in IP phones and gateways differently and there are wide variations between brands, models and software versions. MOS, jitter and packet loss trends are presented graphically on the VoIP Monitor screen . The Diagnostics feature gives additional information about the percentage impact each parameter has on the degradation of speech quality as well as indicating where the network problems lie.
Price Finally, a calculation of all our costs presented in Table 1.
Table.1 - The calculation of price Name Quantity [pc] Price per one pc. [EEK] CISCO827 ADSL Router 2 870 3Com® Switch 4500 PWR 50-Port 2 25 380 ML Novator S570 server computer 2 16 400 Cable Cat 5E 10m 50 300 VoIP Monitor Professional software 1 1 600 101 900 EEK
Conclusion As a conclusion, we want to say that we did our best to make our customer the best offer (in this work, however, the final offer price is not presented as there is need in calculating all our personal work and some money as our benefit from our work). What about this project work, we think that it may be used as a template for future work projects as it shows many aspects in solving some questions in a field of VoIP: building and analyzing the project's scheme, finding suitable equipment and software and etc.
- 10 - Referencies http://www.voip-info.org/ http://www.usr.com/education/voip0.asp http://www.hypercomp.ru/ http://en.wikipedia.org/wiki/Voip http://www.malden.co.uk/voipmonitor.ht m http://www.zoiper.com/ http://www.protocols.com/pbook/VoIPFamily.ht m http://www.asterisk.org/ My laboratory report on subject "IRT0100 Kommunikatsioonivõrkude struktuurid ja teenused"
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