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RTMMoIP. Alfredo Terzoli / Mosioua Tsietsi. PLAN: tiding up…. ENUM: a few more words B/W calculation for VoIP Plan for the rest of the week. DNS-Server. Gateway. Sip server. PSTN to VoIP Call via SIP. Query 1.3.1.9.5.8.6.8.6.4.e164.arpa.?. Response sip:paf@cisco.com. “Call setup”.
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RTMMoIP Alfredo Terzoli / Mosioua Tsietsi
PLAN: tiding up… • ENUM: a few more words • B/W calculation for VoIP • Plan for the rest of the week
DNS-Server Gateway Sip server PSTN to VoIP Call via SIP Query 1.3.1.9.5.8.6.8.6.4.e164.arpa.? Response sip:paf@cisco.com “Call setup” Dial +4686859131 Sip sip:paf@cisco.com (slide by Steven D. Lind,AT&T)
DNS-Server VoIP via SIP to VoIP SIP-Server “ENUM” SIP-Server Gateway Gateway
ENUM ‘strange’ format… • My telephone at home in ENUM format: 5.3.6.3.2.2.6.6.4.7.2.e164.arpa • Why not (it seems more logical!): 6223635.46.27.e164.arpa
ENUM ‘strange’ format • The second way of representing makes explicit in the DNS search the current format of e.164, which might change in future • Using the first one (fully dotted), we isolate ENUM from changes in the e.164 format by the ITU. The only assumption is that e.164 will be numbers, which is safe
VoIP bandwith calculation • Let’s get more specific on B/W usage, for AUDIO transmission • Two families of CODECS: • wave coding • unrestricted, any sound (sound engineering) • vocoding • good for voice, can reach very low bitrates • kbps: kilo bits per second (here kilo=1000, not 1024!)
Voice Synthesisers & Vocoders • A voice synthesiser includes a vocoder of some type, naturally • In fact, using a voice synthesiser one can substantially reduce the bandwidth needed for the transmission of voice. HOW? • (BTW, who knows how MIDI works?)
Bandwidth calculators at: • http://www.newport-networks.com/pages/voip-bandwidth-calculator.html (btw, the white paper distributed to you on b/w calculation comes from newport-networks) • http://www.packetizer.com/voip/diagnostics/bandcalc.html
Reducing overhead • cRTP compact RTP (RFC 2508): • From 40 to 2 to 4 bytes, substantial • Trunking: same packet transport more than one conversation (of course, can be used only if on a trunk)
Main Prac • Building a service in Asterisk • Possible dates: • WEDNESDA 16s: presentation of your idea • TUESDAY 29: demo of the implemented service
A few URLs to follow (by Friday) • http://www.start.com.my/blog/?q=google_talk_vs_skype(blog with comparison of skype and googletalk) • http://gadgets.engadget.com/entry/1234000100055725/ (broadband providers VoIP dilemma) • http://voxilla.com/modules.php?op=modload&name=News&file=article&sid=170&newlang=eng (P2P SIP)