SipTheeSkype - SIP to Skype gateway

long time ago I found a nice Java application that bridge calls between Skype to Sip and vise versa. During the last 2 years i tested few similar solution from NCH with their application name "Uplink" http://www.nch.com.au/skypetosip/index.html
And another application name "PSGW" http://www.rsdevs.com/psgw_sip3.shtml
SipTheeSkype(http://siptheeskype.mhspot.com/) so far is the best one I ever had from the following reasons:
1. SipTheeSkype is an open source product that cost nothing in compare to the other 2 competitors.
2. It has a codec converter to convert SIP RTP audio to compatible Skype PCM audio and Skype PCM audio to SIP RTP audio, while the other 2 solution using virtual audio cable technology for transmitting the voice between Skype to SIP.
3. from testing sip signaling of those applications against few SIP registrar (including Asterisk), SipTheeSkype was the most flexible in interoperability to other systems. I found less signaling bugs in SipTheeSkype than other solutions. I even manged to fix few minor bugs my self by changeing the source code of Mjsip java sip stack.
4. SipTheeSkype have a built in audio prompts for bridging SIP to SKYPE. in the Uplink solution from NCH i needed to install 3 diffrent of thier product to be able to map SIP2Skype calls. first i needed to install their Axon ip pbx with the IVM auto answering and Uplink. Uplink was registered on Axon as extension and the IVM answering attended used for translating sip DTMF to Skype usernames. In SipTheeSkype you have all those functions and more in one product.(need to edit 3 text files and you go)
5. I followed the development of SipTheeSkype for few month and in every new release you get new functions and more configuration options such as new codec (it started with just G711 and today there are ILBEc and GSM support). better DTMF detection with options to change payload (in my case it was important because our system use 96 as DTMF payload while newer standard use 101)....

the only negative thing I can say about SipTheeSkype is its way of configuration by editing few text files while the other solutions are GUI based configuration. to be able to make it work you will need to have some knowledge about SIP signaling as there are many parameters to play with.

Uplink had some major problem of sending sip signaling via Outbound Proxy (SBC) and PSGW which use microsoft FREE RTC sip stack had problems to keep messaging from same source port. If you use Asterisk or any IP pbx on the same location (same LAN) from where you test those solutions than there is no much problems, because you don't need Outbound Proxy or SBC, no any nat problems, and you will probably have Registration attempt every 1 hour.
In My case as public voip service provider the life are not so easy because usually the customer is behind nat and there for need to Register or send Signaling via sbc. The Sbc for keeping the nat port open will ask new Registartion every few seconds - if PSGW can not keep same signaling Source port every Registration will create new contact binding in the Registrar. (its wrong)
IF Uplink can not send correct signaling Via SBC you can not have the solution work with NAT.

MY VERDICT - SipTheeSkype is the winner as it has all in one and more... I had the pleasure to report few bugs to the author and all fixed fast.

Introduction

Hello
My name is Lior Herman from Finland. I decided to open my own blog for sharing my knowledge for VOIP planing, deployment and marketing.
English is not my native language so I am sorry in advanced for any spelling or any mistakes.
Currently I am holding the position of CTO of Suomen Puhelin LTD (Voip service provider in Finland) www.suomenpuhelin.fi
this blog is targeted to technical people who would like to share knowledge helping other to become a successful voip providers.
In my port folio i was involved with consulting and deploying few voip service providers around the world, my expertise cover voip signaling, PSTN signaling and ip networks.
Unlike a lot others voip providers that target their business mainly to residential or international termination, our main target is the business customer. Business customer demand more availability and quality and as service provider we have a small gap for making mistakes, while for residential users it is not so critical.
our main competition is about quality and not cheap calls rate, at least in Finland 2 cent per minutes different will not be the main reason for business user to migrate between operators.
in our location we do what no other big operator do yet and its disconnect the business from its legacy TDM (PSTN) lines and migrate to SIP TRUNK.

I know many that try be voip service provider starting a business in their gardge with some open source solution such as Asterisk and Open Ser and have a great respect for them.
I know Asterisk and other open source solutions, but can say just that:
"To be able to serve corporates customers you need state of the art system in high availability, by the time you will find some bugs and compatibility problems that will need to be resolved and you system supplier will need to do it (unless you know how to change open source). "

I know that a lot of open source freaks will not agree with me and its OK by me as I also use open source solutions myself for some services, but NOT as the core system.

in the present we are working with few market leading solutions like:
Acme packets, Cisco UC5XX, Patton, IPgear/Teles, Nokia, Audiocodes, Siemens......

Tomorrow i will continue is some more details.

Lior Herman