SiptoSis Skype gateway

ckleea

A supplement to our asterisk server

http://www.mhspot.com/sts/siptosis.html

What is the SipToSis Skype Gateway?
SipToSis (Sip to Skype integration software) is Java software that allows you to make and receive Skype calls from your SIP/VOIP telephone adapter, SIP IP phone or Asterisk/SIP PBX. You can also make and receive SIP calls from Skype. It let's you integrate Skype into your SIP VOIP phone system. Basically a Skype/SIP Bridge/Gateway/Proxy/Adapter/Converter. It has a codec converter to convert SIP RTP audio to compatible Skype PCM audio and Skype PCM audio to SIP RTP audio. It performs SIP signaling and Skype call handling to connect with your SIP adapter, Asterisk Server, SIP PBX or SIP VOIP provider.

SipToSis SIP to Skype Gateway Features:

    * Call Skype users using mappings/speed dial or use SkypeOut to make PSTN calls
    * Make SIP calls from Skype using a SIP provider or SIP PBX
    * Skype callers can be directed to the SIP address of your choice
    * SIP callers can be directed to the Skype user of your choice
    * SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks
    * Skype to SIP authentication/denial mappings via incoming Skype User ID
    * SIP DTMF touchtone decoding/encoding via RFC2833, INFO or Inband
    * Skype DTMF touchtone decoding/encoding via Inband
    * Connect Asterisk, FreePBX, Elastix, trixbox, PBX-in-a-Flash, 3CX or other SIP PBX to Skype Users
    * Conference call as of version 20091115 - See FAQ page for compatibility
    * Callback capability as of version 20091115 - See FAQ page for compatibility
    * Skype voicemail retrieval via SIP as of version 20091206
    * Outgoing Skype voicemail support as of version 20091206 - See FAQ page for compatibility
    * Auto play pre-recorded file(s) to SIP and Skype callers
    * SIP caller pin authentication and dialing
    * Skype caller pin authentication and dialing
    * PCMU (u-law)/PCMA (a-law)/G.711/iLBC included codecs and GSM with additional libs
    * Codec interface so you can add other codecs
    * SIP and Skype Hold
    * SkypeOut dialing rules - customize for your location
    * SIP outbound dialing rules - customize for your location
    * STUN support for public IP discovery
    * Call Time limiting ability
    * Usage limiting abilities by used time and unique called number count
    * Can be setup as a multi channel Skype to Asterisk Trunk for multiple simultaneous calls with the stsTrunkBuilder
    * Multiplatform (Windows/Linux/Mac OS X)
    * Run everything on a single computer if you wish
    * Windows users can run it as a service - see Appicus Windows Service Wrapper

SipToSis SIP to Skype Gateway System Requirements:

    * Skype client - See FAQ page for known working versions
    * Sun/Oracle's Java 1.5 or higher (Linux users - Do NOT use openjdk)
    * SIP/VOIP adapter such as a SPA3102, SIP IP Phone, register with a SIP provider or setup an Asterisk/SIP PBX server
    * Skype4Java compatible platform
    * Sufficient bandwidth to support your configuration - Broadband preferred.
    * Skype compatible sound device

This product uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype.

SipToSis is GNU GPL Licensed software.

Online documentation:

    * Frequently Asked Questions/Issues
    * Phone Adapter/Basic Configuration Guide
    * Skype to Asterisk Trunk Setup Guide
    * Troubleshooting

ckleea

Examples are

A call flow example of how this SIP Skype gateway might work with a PBX.

    * SIPCaller --> Asterisk/PBX --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
    * SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> Asterisk --> SIP destination

SkypeSTS is the Skype user that SipToSis is attached to.

A call flow example of how this Skype to SIP Gateway might work with a SIP device.

    * SIPDevice --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
    * SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> SIP Device or SIP Destination

SkypeSTS is the Skype user that SipToSis is attached to.

A call flow example of how this Sip to Skype Gateway might work with a SIP provider.
SipToSis would have to register with the SIP provider to take incoming SIP calls from provider.

    * SIPCaller --> SIPProvider --> SipToSis --> SkypeSTS --> SkypeUser or SkypeOut to PSTN
    * SkypeUser or SkypeIn --> SkypeSTS --> SipToSis --> SIP destination

SkypeSTS is the Skype user that SipToSis is attached to.

Sam

Looks very interesting, anyone tried yet?

ckleea

I have set up 4 channels for use. Only that it is now having a new version that I am not sure how to update.

Working very well indeed.