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Speex Joins Xiph To Bring Free VOIP To The Masses
xercist writes "Xiph.org
has added a new project to their plate of goodies-
Speex.
Speex is an audio codec specifically for, you guessed it, voice.
It has integration with Xiph's
OGG
container, but is mainly being used right now for VOIP.
There is currently an XMMS
plugin
available, and is also supported by
LinPhone,
OpenH323,
and
GnomeMeeting.
Asterisk PBX
is working on adding support.
This is not a new project -- Jean-Marc Valin has been hard at work writing
the codec for quite a while now. However, Jean-Marc is now a full-fledged
member or the Xiph.org team, and in celebration, Speex beta one is being
released.
Xiph.org has brought you
(or is currently working on bringing you)
Vorbis,
Tremor,
Theora,
Tarkin,
Icecast2,
cdparanoia,
now Speex,
and, of course, the
Moaning Goat Meter.
This is a LOT to do, so please
donate
to show your support."
I've been playing around with speex when i was working on an audio conferencing. It's a simple api, and the audio quality comes out okay for voice too. (unless you try sending music through, then it really just craps out)
If only I could get the windows side of the cross-platform audio caputre stuff so nice.
New worlds are not born in the vacuum of abstract
ideas, but in the fight for daily bread --Rudolf Rocke
TCP/IP? That's wasteful...afaik most VoIP use UDP, as TCP carries a much alrger overhead and there's no harm if some UDP packets are lost (hey, you missed 1/50 of voice data. doesn't matter). regarding why internet is preferable, I think it's already been answered...
TCP/IP is not referred as just TCP, it's the name given to the suit of protocols that makes the Internet happen, namely TCP and UDP, and IP, ICMP and others on different layers.
Articulos para gente geek: Poleras, linux, libros y mas
IIRC, NetMeeting allows you to plug in extra codecs. So it shouldn't be too hard to get this working under windows.
The opinions stated herein do not necessarily represent those of anybody at all. Deal with it.
Voice over IP doesn't send voice data over TCP, it uses UDP. UDP isn't complicated at all - it just gives you a way to uniquely identify a machine and say "send this data to it." It doesn't even guarantee delivery of the data. It's probably the best, most accepted way of sending addressed, digital data over wires.
Now, imagine you're a company that's just put an office up. Would you rather install two sets of wires to each desk (ethernet and phone network), one of which requires you to get a licensed contractor in if you need work done on it? Or a single set of wires which can be maintained by the people who run your computers?
Why does everyone insist we need to do absoultely everything over TCP/IP?
Because it allows telcos to switch from circuit switched technology to packet switching. Circuit switching is expensive and complicated; for example, trying to mux/demux multiple channels on a circuit switched line requires some very funky hardware. You can easily mux multiple IP streams with a switch, and route thousands of calls with a single router. Whats not to like about that?
Having said that, most current networks (E.g. the Internet) are built on top of....circuit switched networks. E.g. IP over SONET/SDH. Ah well.
CDex is also very good. It rips from CD directly to OGG, and has all the cool features of other CD rippers. You can create OGG albums with literally one click of a button. Did I mention it's an Open Source windows client? What more could you ask for?
While udp certainly is the right choice for transmitting the actual audio data (low latency etc.) this alone doesn't make a complete telephony protocol.
One standard used often today for call management (listen for incoming calls, register possible recipients etc.) is H323, the one netmeeting and gnomemeeting, among many others, use. Unfortunately H323 does a very bad job when it comes to transmitting data through firewalls, nat-gateways or proxies (typical environment in many companies today) since it contains parts which choose arbitrary high ports for connection. You can work around this by installing e.g. OpenH323 Proxy on your gateway, but usually you'll need your systems administrator to do that - and it is pretty likely that he/she will refuse to do that for security reasons or simply because it can become quite tricky to set up a stable working H323 proxy/gateway (lots of configuration work).
BTW i've heard that some firewall constructors have basically given up on that matter and simply open all ports when they detect some client intends to do netmeeting.
time is a funny concept
Yes your right, Transmision Control Proticol isn't Transmision Control Proticol over Internet Proticol.
User datagram proticol is definatly NOT Transmision control proticol, no matter what it's running over. They have wildly diffrent end goals and thus look nothing alike on the wire.
In the future, if you want to say "Must everything run over the internet" say so, because quite a few TCP networks aern't the "internet". Using a single network for all data has several advantages (especailly when that network is redundantly failsafe, and already built)
I love the fact that a good, Free Software voice codec is out there, and here are my reasons:
1) Ham Radio. The Tucson Amateur Packet Radio organization is working on experimental digitized voice over amateur radio applications, and a couple of venders (mostly Kenwood) are offering radios that have this ability. Right now, TAPR are looking at using DVSI's IMBE vocoder, which is QUITE expensive and VERY not-Free. The availability of a Free codec would greatly improve the availabilty of this protocol.
2) Currently, The Association of Public-Safety Officials (APCO) (the folks who define the specs for the radios used by police, fire, and government) have defined the current digital trunked radio standard, APCO Project 25 as using DVSI's IMBE vocoder. While this is licensed under a Reasonable And Non-Discrimitory license, if you want to license the IMBE vocoder for a P-25 project, you will cough up US$100,000.00 for the privilege (I know firsthand, as the company I work for has done this). Uniden, Radio Shack, and other scanner companies are looking into putting this into their scanners, so they have had to cough it up as well. A Free vocoder would allow anybody to build a product with this capability in it - you could even use a scanner and your sound card to decode the Phase 1 C4FM format signals.
Like so many other things, a Free Software tool to do these things would greatly accelerate the industry. I hope Xiph does well.
www.eFax.com are spammers
(I am the Speex author) There are already at least two Windows front-ends: here and here. There may be others I'm not aware of. Note that I haven't developed of tested any of these since I don't use Windows.
Opus: the Swiss army knife of audio codec