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bounty skype

Created by: caryon,Last modification on Mon 15 of Sep, 2008 [10:32 UTC] by pkolmann

Bounty to be paid for Skype functionality for Asterisk


Manager: ason Toy
Bounty: aprox. $1,645 USD ( $200 + $200 + $50 + $30 + GBP100 + $20 + EUR 20 + $1,000 USD)
Date opened: November 9. 2004, Updated May 31, 2007
Status: Open

Contributors:
- Dan - $1,000 USD contact: dan
- Jason Toy - 200 USD contact: jtoy
- MuppetMaster - 200 USD contact MuppetMaster
- Fran - 50 USD contact (please leave a contact to take you seriously)
- Sjobeck - $30 USD
- BCN Telecommunications - 100 GBP contact: nickatbcndashteledotcom
- Adrien 20 USD contact: modulis
- Conception blak kat 20 CA
- Talamona 20 EUR
- Mjaavatt 20 USD
(- caryon - canceled - there is a commercial one I can use)

We will need a trustee if someone is only willing to pay anonymous.

Details:
There a two possible ways to do this.
Create a Asterisk extension,
1. with Skype low level functionality (reverse engineering)
2. on top of the documented Skype API (without running a skype Client)

Call quality should be equal to skype, also must include at least a linear pcm converter

In any case it has to work on linux as an additional module for asterisk.
Multiple account-login should be possible.

Note: Please consider the following limitations of the interface to the Public Skype API:
1. The Skype 'engine' requires X-windows to be running
2. Currently the audio stream cannot be directed at anything other than /dev/dsp.
3. Only one (asterisk) client can use it at a time.
4. Interface to the engine is via a specific version of DBUS

2 and 4 are planned to be changed in future, but in order to write a channel driver all of the above need to change. If you want Skype to consider changing it's API, then contribute here: https://developer.skype.com/wiki/NakedSkype

== compilation of information ==
  • Skype API information Skype-devzone
    • Skype for Linux public API is here. It's based on D-BUS or X11 Transport Skype API on Linux
  • Analyzes of the Skype protocol from the Columbia University Computer Science Department PDF
  • Method 2 isn't possible as mentioned above. The API is only provided by the running skype client via dbus. You can controll some client stuff via this dbus-api, you could write a chan_skype to place outgoing calls (shouldn't be that hard) but in order to get the audio stream you must either capture the stream written to /dev/dsp (skype only handles oss) with overwritten syscalls (read,write) and LD_PRELOAD or use an appropiate dummy-sounddevice. The API documentations explains that you can change the audio devices with SET AUDIO_IN/OUT, but that doesn't work at all. While this all requires a client running, the only possibility to apply for the bounty seems to reverse engineer the protocol and implement that into a channel driver.

== background information ==

== user thoughts ==

== Original Skype Linux API ==

== Try ChanSkype ==
  • Although not open source, ChanSkype is an option for those who are willing to pay a license per incoming or outgoing channel.



Comments

Comments Filter
222

333

by p__m, Friday 08 of December, 2006 [18:31:24 UTC]
I recently had the idea to write the following skype daemon:

- the daemon starts on demand one (or more if needed) xvfbs that are a kind of null-x servers
- for each client session, the deamon starts one skype process. the process is set up to use the xvfb as x server
- a library is preloaded to the skype program that fakes a /dev/dsp (like esddsp, artsdsp or alsadsp do)
- the skype process is controlled over the skype api
- clients would communicate with the daemon via IPC
- there would be a library to communicate with the daemon used for both signalling and sound. an asterisk module would use only this library

Technically, although not ellegant, it would be possible i guess and it would work. The only problem is it would break the following eula points:

- we are not allowed to distribute software that uses the skype api (providing source code also probably means distributing i guess) unless skype allows us to
- we are not allowed to hide the damn gui
222

333It's been done...

by timhunt, Tuesday 31 of October, 2006 [13:27:49 UTC]
Take a look at http://www.chanskype.com.

It's not elegant, and each instance requires a Skype session, but multiple Skype sessions are supported using VNC.

Be warned that (as of 10/31/2006) Version 1.2.X does not install cleanly on a Trixbox type configuration but it can be made to work.
222

333Skype2iax

by lanonyme, Friday 28 of April, 2006 [14:20:12 UTC]
Hi,

I'm currently developping it.
First beta will be released very soon.

Functionnalities are :
- Easy way to create Skype API, IAX and both protocols based apps in java.
- Skype To Asterisk call forwarding.


This solution uses a modified version of JSkype lib and IAXClient lib.

links :
- http://iax.skype2rtc.info/ (Temporary)
- http://groups.google.com/group/skype2iax 


For more information, please, contact us.
222

333Skype device hints

by burdick0608, Tuesday 31 of January, 2006 [19:09:27 UTC]
Now that there are skype devices hitting the market, maybe they could be a starting point. The netgear skype wifi phone (needs no PC) might just run linux itself?
http://tools.netgear.com/skype/
222

333Skype & Asterisk integration proposal

by vitaly_repin, Friday 23 of December, 2005 [09:19:55 UTC]
There is a following possibility: skype (several instances) installed on the server-side and connection between this skype and asterisk. It will be possible to call skype users from any device connected to asterisk and to use SkypeOut in the same manner.

Benefints: 1. Skype installed on only one computer (some kind of DMZ) - security improvements. 2. Company shouldn't upgrade end-user devices to use skype. 3. It is possible to make some kind of accounting (of skype usage).

Is this capability intrested, guys? Are you ready to pay for this project?
222

333++Skype

by smegma, Tuesday 15 of November, 2005 [20:47:08 UTC]
How about this:

http://www.icebrains-soft.com/getting_started_with_the_skype_library
222

333Supply skype voip phone & gateway - by alfatec

by alfatec, Thursday 10 of November, 2005 [06:33:31 UTC]
Dear Sir or Madam:

This is Kevin Zhou of AlfaTec. We are the telecom hardware production company in China. We are also the member of Sino-German IT Cooperation.

We have designed and produced USB handsets and gateway for working with Skype and other web phones. We are also doing VOIP product OEMs for other telecom carriers. We also produce SIP Phones and voip gateway for Chinese telecom carriers (www.chinatelecom.com.cn, http://www.chinanetcom.com.cn).

We are looking for partners and agents to distribute our products in the wider range, please contact us if you are interest in our products and willing to establish service cooperation with AlfaTec.

Below are the some of features of the USB products are well received by our customers.
You may visit http://www.alfatec.com.cn for more information.

USB Phone (with LCD) :A-200
This fully-featured USB VOIP Phone allows you to talk to friends or colleagues just like a normal telephone. Used in conjunction with web phone software (like skype), it offers you a complete free Internet telephony solution. The AU100 features simple plug-and play installation, integrated keypad, echo cancellation and noise reduction.

USB VOIP Phone Benefits include:
• USB 1.1 support
• LCD
• 16 bit sound card inside
• Superior sound quality (improved sound speaker)
• Turns your PC into a real Phone, talk to friends or colleagues just like a normal phone
• SKYPE* ready – talk for free with other Skype users
• Easy to install USB plug-and-play technology
• Allows PC-to-PC and PC-to-phone operation
• Echo cancellation and noise reduction
• Supports Skype, NetMeeting, Messenger, Net2phone, Sjphone, StanaPhone, X-Lite, MSN, MediaRing, and Dialpad
• Supports H.323, MGCP, SIP protocol
• Host powered – no external AC adapter required
• Full control and functionality over Skype software
• HID support
• Small and slender design - Reducing desk clutter
• Ring incoming call
• Adjustable volume for handset
• Enterprise class solutions at consumer prices
• Supports Windows® 98SE/ME/2000/XP. No driver needed for Win98 SE/Win ME/ Win 2000/Win XP/ Mac OS/Mac OS X
USB Skype Gateway:A-600
• Continue to make and receive regular calls as you normally do
• Make and receive Skype™ calls using your standard telephone
• Forward Skype calls to your mobile phone
• Make Skype calls from your mobile phone even when you are away from your computer
• Switch between a Skype call and a regular phone call

AlfaTec USB phones are manufactured in a state of the art production facility in our factories in Shenzhen under meticulous quality control. Join us now, grow your business with us.

Best regards,
Kevin Zhou
International Channel Sales
AlfaTec Technology Co., Ltd. (A member of Sino-German IT Cooperation)
Email: Kevin@alfatec.com.cn
sales@alfatec.com.cn
http://www.alfatec.com.cn
(Website in Beijing China, mirror site in H.K)
http://www.sg-itc.com.cn
Tel: (86 10) 5166 2853, 51662855, 51662856
Fax: (86 10) 6508 1026
Skype: alfatec-kevin
Address: Beijing Office (Headquarter): Suite 904, E-Tower, No.12, Guanghualu, Chaoyang District, Beijing 100020, P. R. China

Factory in Beijing: Address: 3C, Beijing Shi men economic development zone, Men Tou Gou District, Beijing, P.R. China 100010
Factory in Shenzhen: Address: 3F, Area 2, Block1 Xili Pingshan Honghualing Industrial Estate, Shenzhen .P.R.C Post code:518055
VOIP USB phone for Skype, Messenger, Net2phone, Sjphone, StanaPhone, X-Lite, MSN, MediaRing, and Dialpad.
www.alfatec.com.cn Your VOIP Hardware Partner!



222

333Microsoft VoIP

by turby, Monday 05 of September, 2005 [20:30:08 UTC]
there is second thing for development in near future - microsoft VoIP solution for windows...
222

333Solved: Skype Gateway

by daniel@jajah.com, Friday 24 of June, 2005 [12:50:44 UTC]
The softphone http://www.jajah.com offers already jajah to Skype calling without having Skype installed. It supports SIP, IAX and POTS termination, too.
222

333I'd like to take a crack at this

by marcus_macd, Sunday 22 of May, 2005 [08:18:51 UTC]
Hey i think this is possiable someone contact me

marcus_macd@hotmail.com