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.
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4 comments:
Really appreciate that you open up your knowledge to the public. Really enjoyed your comment about the siptheeskype, for I am just start looking to play with it a little.
I am in California, US. Used to work for an ITSP, also focused on business customers. And true, they demand quality more than anything else. I see also you have the ability of coding same languages, that's a huge plus. I am still struggling with the entry level programming here. Hope i will be catching up soon.
Please, How can I get the source of this application. I would like to review it.
please give me a hint, the webpage said they are open source, but cannot find it.
mike
Hello Mike
the source code with instruction should be in the zip file with the application itself.
Lior
Product is now known as SipToSis and is still available at the site linked in the original article. I agree, that the call quality is far superior to the other products out there.
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