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Wideband Audio Conferencing with Asterisk

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Wideband Audio Conferencing with Asterisk Wideband Audio Conferencing with Asterisk Presentation Transcript

  • AG Projects ICE: the ultimate way of beating NAT in SIP Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Saúl Ibarra Corretgé | AG Projects Because G711 is not enough AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Index ● What is Wideband voice? ● Should I use Wideband anyway? ● Asterisk wideband capabilities ● Conference calls with Asterisk ● Testing and results AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts What is Wideband voice? ● Higher quality voice ● Higher detailed voice ● Richer sound Nothing new! G722 is from 1988! AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts What is Wideband voice? (II) ● Human voice ranges from 30 to 18000 Hz ● The more frequencies we transmit the richer the voice is AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts What is Wideband voice? (III) ● Nyquist sampling theorem ● If we want to transmit X amount of frequencies they need to be sampled at 2X sample rate ● G711 ● ~50 – 4000 Hz -> 8000 Hz sample rate ● G722 ● ~0 – 7000 Hz -> 16000 Hz sample rate AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts What is Wideband voice? (IV) How can G711 and G722 use same bitrate (64 kbps) then? ● G711 ● Uses PCM. 8 bits per sample * 8000 samples = 64 kbps ● G722 ● Uses SB-ADPCM – 48 kbps for the lower band – 16 kbps for the higher band AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts What is Wideband voice? (V) AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Should I use Wideband anyway? Short answer: yes. Long(er) answer: yes, of course. ● “Cleaner” sound ● Easier way to identify voices ● Clearer diference between close sounds: “sailing” vs “failing”, etc. ● Just don't say “what?” ever again. 25% of calls at SIP2SIP.info use wideband Only 2% did use wideband same period last year AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Asterisk codec capabilities Narrowband codecs Wideband codecs ● G711 ● G722 ● GSM ● G722.1 (Siren 7 and Siren ● G729 14) ● iLBC ● Speex 16 KHz (new in 1.8!) ● G723.1 ● G719 passthrough (new in 1.8!) ● G726 ● ... AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Conferencing Multiple callers involved in a single call AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Conferencing (II) ● Client side ● Server side ● Hosted service AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Asterisk conferencing capabilities Asterisk supports several channel independent applications for conferencing: ● MeetMe ● It's been around since forever ● Mixing is done in DAHDI really ● No wideband support (mixing is done in DAHDI at 8 Khz) AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Asterisk conferencing capabilities (II) ● ConfBridge ● New as of Asterisk 1.6.2 ● Uses “new” Brdging API ● No DAHDI needed ● Wideband capable! ● Can do mixing at 16 Khz or 8 Khz, but not both ● Not as many options as MeetMe... yet! AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Asterisk conferencing capabilities (III) ● AppKonference ● Third party application ● Fork of AppConference, around since Asterisk 1.0.x ● No DAHDI required ● Wideband capable! ● Video capable! ● Not working with Asterisk 1.8 yet AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts A note on timing sources ● Full explanation on doc/timing.txt ● Asterisk supports several sources which can be used to provide internal timing ● res_timing_timerfd – Only available on Linux systems with Kernel >= 2.6.25 & glibc >= 2.8 – Very reliable source of timing ● res_timing_kqueue – Only available on BSD systems (yes, also works on the Mac) ● res_timing_dahdi – Uses DAHDI to provide timing ● res_timing_pthread – Uses the pthread library to provide timing – Least efective, but more portable AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts ConfBridge exten => _*7XXXX,1,NoOp(Entering conference ${EXTEN:4}) same => n,Answer same => n,ConfBridge(${EXTEN:4},M) ● No confguration fle ● Smart bridging: 2 party and multiparty mixing ● Be careful, you can't mix diferent sample rates ● Inbound codec can't be forced ● Not as complete as MeetMe... ● ... I'm sure patches would be welcome :-) AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts AppKonference exten => _*7XXXX,1,NoOp(Entering conference ${EXTEN:4}) same => n,Answer same => n,Konference(${EXTEN:4},H) ● No confguration fle ● Inbound codec can't be forced ● Minimize encoding/decoding ● One speaker: frames sent directly to each participant. Frames transcoded once per codec type ● Two speakers: each speaker gets the other speakers frames. The two speakers frames are then mixed and transcoded once per codec type AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Testing ● See how diferent options perform in terms of CPU usage ● AppKonference ● ConfBridge (with TimerFD timing) ● ConfBridge (with Pthreads timing) ● Test scenarios (always 50 users) ● 1 speaker ● 2 speakers ● Everyone speaking, madness! AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Tools ● pcapsipdump: capture SIP + RTP in PCAP format discarding everything else ● Wireshark: edit captured PCAP ● SIPp: SIP + RTP trafc generation ● sysstat: system stats generation ● OpenOfce: graphics. It sucks, btw. ● Human hear ● Hardware ● Code2Duo desktop computer with 8GB of DDR3 RAM ● Gigabit Ethernet AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Tools: Asterisk versions ● ConfBridge tests ● Asterisk SVN branch 1.8 r292230 ● AppKonference tests ● Asterisk 1.6.2.14-rc1 AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Results: 1 Speaker Single Speaker 45 40 35 30 25 1 Speaker (AppKonference) 1 Speaker (ConfBridge + TimerFD) 1 Speaker (ConfBridge + Pthreads) 20 15 10 5 0 1 2 3 4 5 6 7 8 9 101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960 AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Results: 2 Speakers 2 Speakers 70 60 50 40 2 Speakers (ConfBridge + Pthreads) 2 Speakers (ConfBridge + TimerFD) 2 Speakers (AppKonference) 30 20 10 0 2 4 6 8 10 12 14 16 18 20 22 24 26 28 30 32 34 36 38 40 42 44 46 48 50 52 54 56 58 60 1 3 5 7 9 11 13 15 17 19 21 23 25 27 29 31 33 35 37 39 41 43 45 47 49 51 53 55 57 59 AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Results: 50 Speakers! 50 Speakers 80 70 60 50 50 Speakers (ConfBridge + Pthreads) 40 50 Speakers (ConfBridge + TimerFD) 50 Speakers (AppKonference) 30 20 10 0 2 4 6 8 10 12 14 16 18 20 22 24 26 28 30 32 34 36 38 40 42 44 46 48 50 52 54 56 58 60 1 3 5 7 9 11 13 15 17 19 21 23 25 27 29 31 33 35 37 39 41 43 45 47 49 51 53 55 57 59 AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Results analysis ● AppKonference had more audio cuts (according to Human Hear TM) ● Short audio-loss when lots of calls were starting (ConfBridge) ● Overall TimerFD performed better than Pthread ● AppKonference was tested on a diferent release version ● With 50 speakers AppKonference produces no audio at all ● ConfBridge doesn't have as many features as good old MeetMe AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Recap ● Wideband can make our conference calls more understandable ● With G722 we use same bandwidth as G711 ● Asterisk provides all the necesary tools for a nice and wideband conference call experience ● ConfBridge looks like the way to go ● All hardware devices involved need to support wideband AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Spam! ● Join the VoIP Users Conference! ● Every friday at 12:00 EDT ● Highly skilled speakers talking about VoIP ● Anyone can join the conference! ● In G722, of course! http://vuc.me AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Questions? AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts BYE BYE sip:audience@astricon.net SIP/2.0 Via: SIP/2.0/UDP 192.168.99.23:49919;rport;branch=z9hG4bKPjDb30Dx0sH-ozn9QB.cCCboyU.atR97aM Max-Forwards: 70 From: "saghul" <sip:saul@ag-projects.com>;tag=UCpGKVZbQQx7BUKYtiuPEX668oa9jaU7 To: <sip:audience@astricon.net>;tag=as59aef35c Call-ID: DEWDfu63OACwYeQk7MrhmRhRq.1cqqis CSeq: 10633 BYE Route: <sip:81.23.228.129;lr;ftag=UCpGKVZbQQx7BUKYtiuPEX668oa9jaU7;did=641.a8a9c553> User-Agent: blink-0.20.2 Content-Length: 0 @saghul saul@ag-projects.com sip:saul@ag-projects.com AstriCon 2010
  • AG Projects Wideband Audio Conferencing with Asterisk The SIP Infrastructure Experts Images http://www.fickr.com/photos/kigs/4991332361/sizes/l/in/photost ream/ http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/pho nes/ps379/ps8537/prod_white_paper0900aecd806fa57a.html http://www.fickr.com/photos/timdorr/2737609108/sizes/z/in/pho tostream/ http://www.trennum.net/hullabaloo/ AstriCon 2010