M151. Analiza practica a solutiilor de VoIP
1.Tema proiectului:
Tehnologia VoIP nu este o tehnologie foarte noua, insa doar recent a devenit foarte raspandita si destul de reliable pentru a fi folosita ca solutie de business. Comparativ cu PSTN, tehnologia VoIP ofera mai multe avantaje, intre care cele mai importante sunt costul redus si posibilitatea mai multor convorbiri simultane.
2.Obiective:
Se va incepe prin analiza protocoalelor de semnalizare folosite intr-un sistem VoIP: SIP, H.323, a protocoalelor de transport (RTP, RTSP) precum si a codec-urilor folosite pentru compresia pachetelor (G.711, G.729, G.723.1, etc).
Simularea unei infrastructuri ce permite efectuarea apelurilor si analiza intarzierilor in functie de tehnologiile folosite.
3.Bibliografie:
[1] Switching to VoIP, Theodore Wallingford, O’Reilly Media
[2] http://www.iptel.org/doc
[3] http://www.sipcenter.com/
[4] http://www.voip-info.org/
4.Detalii de desfasurare:
- Coordonator proiect: conf. dr. Ing. Razvan Rughinis
- Echipa: Ruxandra Burtica
- Cunostinte necesare: retelistica, administrare servere
- Sala: EG106b
- Program: 6 ore pe saptamana, doua semestre
5. Rezultatele primului semestru
VoIP Solutions
After this first semester of research, I can say that one can really improve the bandwidth consumption without having a bad sound quality, by replacing G.711 with G.729, by looking at the graph generated and the MOS for both of the codecs. But I can only say this based on some tests and with one call at a time. For the weeks to come, I plan to implement the automation of calls, which will help in generating real-life scenarios, like peak intervals.
The automation can be done in different ways in Asterisk, including the method of using and generating .call files, a method that I tried half through. Besides automating the calls, I shall use a softPhone that accepts multiple calls in the same time (X-Lite doesn’t have this feature) and check if it is possible to create a script that automatically answers the generated calls.
After a good batch for executing the tests is ready, it can be useful for VoIP companies or companies that implement VoIP as a service for business purposes. It is very useful to have a platform for testing bandwidth consumption, matching between codecs, voice quality and other aspects, before making a decision on a bigger implementation.
Another future work that I want to implement is having 2 Asterisk servers communicating, instead of one that passes calls from one user to another. The 2 servers will communicate using the IAX (Inter-Asterisk eXchange) protocol. A left aside this semester is implementing H.323 communication, because of the clients (most of them only just support SIP with G.711 and GSM, most don’t support multiple calls, some don’t support both protocols).
