Slashdot Mirror

Some comments on the following;

Furthermore, MANY VoIP routing solutions are entirely dependent on the H.323 or H.323v2 pseudo-standard, which uses UDP - an inherently unreliable transport - to transmit and recieve call routing information on a per-system per-port level. H.323v2 is also bandwidth intensive. And I have yet to see a system that can interconnect H.323v2 with SS7, which is the absolute standard for all telephone call routing, as defined by Telcordia/BellCore. The Lucent 5ESS runs SS7 for routing, I shouldn't need to say any more than that. This may have changed in the time that I have been out of VoIP, but I doubt it

H.323 is indeed bandwidth intensive. It takes a number of nailed up TCP(!) connections to maintain a 323 call. However, H.323 v1 and v2 do not support call connects over UDP. It wasn't until v3 that UDP based communication came in to play. SIP supports both TCP and UDP connections. It also greatly reduces the number messages needed to set up a connection between endpoints.
If you look at SIP and MGCP (IETF equivalents to H.323) there are a number of efforts to put in SS7 interoperability. ISUP traffic over SIP is in place today. And there are products available to do this. (See SIP-T which is SIP-Telephony interoperability). I'm not a huge fan but you might also want to look at http://www.softswitch.org/. Their are a number of companies attempting to build class 5 switches based on VoIP technology (see IPVerse, XyBridge, Lucent, etc).
If you're interested in AIN functionality over IP you can try the Generic Data Interface (Bellcore SR-3389). This is all North American. For ITU/ETSI you'll have to go do the legwork, I'm not sure there's anything specified.

To sum up, if any of you out there are actually interested in the current market for VoIP and the capabilities, go do your own legwork. Don't listen to one isolated experience on /.. The large majority of people in telecoms know jack shit about internet technology, and the same goes for internet folks coming in to telecoms. I'm not claiming that you don't know what's up RISCy, just that 1998 is eons ago in this industry and things are changing rapidly. The VoIP industry is blowing up right now and there's a lot of heavy shit going down. I highly recommend it to anyone interested in protocols. Here are some links:
SIP home page, great FAQ and Links.
Alright H.323 starter
Open source H.323 effort.
SIP-T starter.

For Real world examples try Dialpad.com, www.talkopia.com, etc, etc.

• Crossplatform - runs on Windows and *nix (Im not sure about Max or Bez et al)
• It has been tested and interoperates with more H323 stacks than you knew existed - Radware, Cisco...
• Free Software - which is the point of this article, right?

• Very complex ... just take a look at the codebase
• Control data is transmitted in binary form - most widely used protocols are based on ASCII (FTP, HTTP, SMTP)
• It uses a port assignment process which is virtually impossible to use through a NAT firewall.

i was implementing a VoIP solution for a company a few weeks ago where linux based kiosk terminals where euquiped with phones that should be able to make VoIP calls to a central callcenter including video. i looked at the existing software. H323 seems to be the way of the futurere and there are already H323 based solutions on the way. openh323.org. even thought they where not stable enought for my needs yet. but at least the demo appication (voxialla) was able to interoperate with the M$ netmeeting shit. video transmition with H263 codec (for low bandwith) is also on the way. for my solution i decided to use quicknet telephony cards (it greatly enhances the telphony experience if you have a real phone connected to your computer which can also ring and is independent of the sound card). those have a DSP on board which does voice compression accourding to the most important standads. it has a GPL'ed kernel driver. (the only downside is the DSP code itself is not open but that is not that much of a problem).

i decided to just adopt the demo code that came with the quicknet cards for my appolication since it was more stable then the H323 things. (it is easy since the compression is already done on the card). for the video thing i used a parallel netscape server push with 1 picture every 3 or 4 sekunds 160x120 (about 2k Byte) in size). greetings mond.

There's an interesting solution I heard of recently : OpenH323 is an open source implementation of the H323 specifications (it covers several audio and video conferencing protocols and codecs).

Voice over IP isn't really mature yet, but there are usable implementations (both commercial and free). The main competing protocols are the Session Initiation Protocol (SIP) created by the IETF, and H.323, created by ISO. SIP is much simpler and easier to understand, but H.323 has a lead in deployment.
Take a look at:
Internet Telephony for a very good overview of the issues and technology,
The Session Initiation Protocol page for SIP info and comparisons with H.323,
The OpenH323 Project for a free implementation of H.323, and
Vovida for a set of free implementations under development for both protocols. On the commercial side, Computer Telephony magazine has loads of information about VoIP and related topics, including a feature article this month about SIP.

Combine that with the software at www.openh323.org you can do excellent quality VoIP calls TODAY with open source software all the way to the metal.

Get more info at Quicknet's web site.

The standard is called H.323; you can get an Open Source, patent-free implementation from the OpenH323 project.

Remember - that's one way! You need to add the reverse direction, and then add the packet overhead of RTP (real time protocol) and UDP, plus the control signals. You can do great IP calls and use about 16kbps or less. I know. I do it all the time!

Our products (PCI/ISA/PCMCIA) allow you to connect a normal phone to your PC (and provide dial tone, DTMF, ringing, etc) and provide the compression codecs. We have open source linux drivers, and are in the kernel as of 2.2.14. For more information, write me or see our web site: Quicknet Technologies, Inc.

Check out the OpenH323 Project and the OpenPhone Project for more information.

These applications run on linux (and Win32) and work very well, even over dialup connections. If you use hardware to do the audio compression you can use a normal phone and get MUCH better performance. Disclaimer: my company, Quicknet Technologies, Inc. makes this kind of hardware. We even have open source drivers, and are in the kernel as of 2.2.14! We're hiring too!

Internet Telephony is the next big thing - and linux will be a major platform for it.

The GPL issue may quite likely be valid in the future as we consider incorporating pieces of software such as that from www.openh323.org, an open VoIP stack (actually under the Mozilla license). Any such source modifications will most certainly be made available.

As far as the web page, that's by a marketing department which wouldn't no what GPL is (-.

The concept is called "protocol encapsulation." As in H.323 videoconferencing protocol over HTTP. Since we also must do videoconferencing with our customers, we need a configuration that works over the Internet as well as the LAN, and our firewalls are configured such that HTTP is the viable solution.

As for file transfers, HTTP-DAV is nice because it allows totally clueless end-users to post documents via the web browser, rather than use some other application to handle file transfer. Since they are already using the browser as the interface for just about everything else, allowing them to post documents via the browser reduces end-user training requirements.

I'm not certain what you mean by using protocols that actually suit what I'm doing, as these two protocols (H.323, HTTP-DAV) are being used exactly as the designers intended.

The OpenPhone Project aims to make it easier for this kind of software to get built. Other good links include:

OpenH323 Project

Linux Telephony

Voxilla - More Linux Telephony

Here is a random post by Craig on the OpenH323 mailing lists.

If you're interested in telephony, check out

• http://www.openh323.org -- desktop devices
• http://www.linuxtelephony.com -- phone hardware

- |Daryll

This is a project that I am working on, yes the 3com, as you can read works under Linux ... I have a good old Bt848 TV capture card ... if building the Video4Linux package though - I say beware - if you have a tuner card - be very scared. (grin) Now as for Netmeeting Go to http://www.openh323.org/ There you will find out that Netmeeting and VDOphone use a standard called H323 - which is the good old ISDN videoconferencing standard - for us people who remember ISDN ;-) ... it works with a 28.8 modem - but not great ... Mind you if you want to download the document, you should go to the library and get the ITU standards book because it will cost you money to download it :/

But there is hope -- as for a client it would appear that there is one that is being developed called Voxilla - check it out - I am guessing that they would be more than happy to get ahold of some coders... I think that having a vdo phone would be a wonderful addition to any Linux (granted enough bandwidth).

... and it is called H.323. Check out http://www.openh323.org/.
---

http://www.openH323.org/

Juan

An open source H323 project recently got started:

http://www.OpenH323.org/

Not much there yet, but it looks promising.