bounty skype

This is outdated and just here for historical purpose, there are several ways to integrate skype into asterisk, including an official chan_skype from Digium/Skype

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 SipToSis - Open source ==

This is outdated and just here for historical purpose, there are several ways to integrate skype into asterisk, including an official chan_skype from Digium/Skype

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 SipToSis - Open source ==

Created by: caryon, Last modification: Tue 12 of Jun, 2012 (04:41 UTC) by admin
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