Wideband Audio Conferencing with Asterisk

AG Projects ICE: the ultimate way of beating NAT in SIPThe SIP Infrastructure Experts

AstriCon 2010

Wideband Audio Conferencing with AsteriskSaúl Ibarra Corretgé | AG Projects

Because G711 is not enough

AG Projects Wideband Audio Conferencing with AsteriskThe SIP Infrastructure Experts

AstriCon 2010

Index

● What is Wideband voice?● Should I use Wideband anyway?● Asterisk wideband capabilities● Conference calls with Asterisk● Testing and results


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What is Wideband voice?

● Higher quality voice● Higher detailed voice● Richer sound

Nothing new! G722 is from 1988!


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What is Wideband voice? (II)

● Human voice ranges from 30 to 18000 Hz● The more frequencies we transmit the

richer the voice is


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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


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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


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What is Wideband voice? (V)


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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


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Asterisk codec capabilities

Narrowband codecs● G711● GSM● G729● iLBC● G723.1● G726● ...

Wideband codecs● G722● G722.1 (Siren 7 and Siren

14)● Speex 16 KHz (new in 1.8!)● G719 passthrough (new in

1.8!)


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Conferencing

Multiple callers involved in a single call


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Conferencing (II)

● Client side

● Server side

● Hosted service


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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)


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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!


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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


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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


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ConfBridge

● 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 :-)

exten => _*7XXXX,1,NoOp(Entering conference ${EXTEN:4})same => n,Answersame => n,ConfBridge(${EXTEN:4},M)


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AppKonference

● 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

exten => _*7XXXX,1,NoOp(Entering conference ${EXTEN:4})same => n,Answersame => n,Konference(${EXTEN:4},H)


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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!


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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


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Tools: Asterisk versions

● ConfBridge tests● Asterisk SVN branch 1.8 r292230

● AppKonference tests● Asterisk 1.6.2.14-rc1


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Results: 1 Speaker

1 2 3 4 5 6 7 8 9 101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960

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5

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Single Speaker

1 Speaker (AppKonference)1 Speaker (ConfBridge + TimerFD)1 Speaker (ConfBridge + Pthreads)


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Results: 2 Speakers

12

34

56

78

910

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1314

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1920

2122

2324

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2728

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3132

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2 Speakers

2 Speakers (ConfBridge + Pthreads)2 Speakers (ConfBridge + TimerFD)2 Speakers (AppKonference)


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Results: 50 Speakers!

12

34

56

78

910

1112

1314

1516

1718

1920

2122

2324

2526

2728

2930

3132

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0

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50 Speakers

50 Speakers (ConfBridge + Pthreads)50 Speakers (ConfBridge + TimerFD)50 Speakers (AppKonference)


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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


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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


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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


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Questions?


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BYEBYE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.99.23:49919;rport;branch=z9hG4bKPjDb30Dx0sH-ozn9QB.cCCboyU.atR97aMMax-Forwards: 70From: "saghul" <sip:[email protected]>;tag=UCpGKVZbQQx7BUKYtiuPEX668oa9jaU7To: <sip:[email protected]>;tag=as59aef35cCall-ID: DEWDfu63OACwYeQk7MrhmRhRq.1cqqisCSeq: 10633 BYERoute: <sip:81.23.228.129;lr;ftag=UCpGKVZbQQx7BUKYtiuPEX668oa9jaU7;did=641.a8a9c553>User-Agent: blink-0.20.2Content-Length: 0

[email protected]

sip:[email protected]

@saghul


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Images

http://www.fickr.com/photos/kigs/4991332361/sizes/l/in/photostream/

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps8537/prod_white_paper0900aecd806fa57a.html

http://www.fickr.com/photos/timdorr/2737609108/sizes/z/in/photostream/

http://www.trennum.net/hullabaloo/

Wideband Audio Conferencing with Asterisk

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