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Where's the documentation or manual?
There isn't any "documentation" or manual. Setup can be simple or complicated depending on your capabilities and
what you are trying to accomplish. There are many comments in the configuration files to help with settings.
I have created some setup guides to help with basic setup:
What is the stsTrunkBuilder and why do I need it?
The stsTrunkBuilder is a configuration helper utility that creates configuration profiles
for running multiple SipToSis and multiple Skype instances.
It creates channel startup/shutdown scripts and Asterisk compatible peer information.
It creates the chaining configuration of multiple Skype accounts so you can advertise
a single Skype Userid and still take multiple incoming Skype calls.
stsTrunkBuilder makes multi channel setup and making changes much easier.
If you intend to setup only a single SipToSis channel then you do not need it.
Is it possible to use the same Skype account for multiple channels incoming and outgoing?
Yes, it is possible using the stsTrunkBuilder with some side effects.
If there is a ringing incoming Skype call, no new calls can be made until that call is handled.
Calls already in progress are not affected.
The incoming calls should be answered immediately to minimize the issue. In a PBX such as Asterisk you can
set it up to answer then play a message to the caller while the call is being routed to the true destination.
Another solution would be to use two Skype accounts, one for incoming and one for outgoing. This way
outgoing calls will never be affected by the Skype incoming calls. To make either solution work requires
some additional programs and setup.
For Windows: Setup stsProxy and the Appicus service wrapper.
For Linux: Setup stsProxy.
If you don't need incoming calls and just need outgoing, no additional programs are required.
What SIP phone adapters can I use with this SIP Skype Gateway?
There are many SIP phone adapter devices that will work.
The Linksys Sipura devices have a flexible dial plan
that allows setup without a PBX.
I use the Linksys SPA3102 since I also use a PSTN line.
The Fritz!box is another powerful device.
If you aren't using a SIP provider then just
about any SIP device should work.
It also works with many softphones such as X-Lite, XPRO and Linphone.
For an ATA to directly connect to SipToSis without a PBX using the same line
as a SIP service provider you must meet the following requirements:
- The ATA must not be configured with an outbound proxy. (Some SIP providers require this)
- The ATA must be able to route calls via the dial plan. Fritz!box and many Linksys/Cisco devices can do this easily.
If your ATA or SIP provider does not meet the above requirements,
then use of the ATA's second line is required or the use of a PBX.
If you don't use a SIP service provider, you can configure your ATA to use SipToSis as
the actual provider.
You can get SIP/ATA devices here.
How do I setup a PBX such as Asterisk to do Skype trunking/connect Asterisk to Skype?
You can setup a PBX to make multiple active Skype calls by setting up multiple instances of Skype and SipToSis.
Each instance/channel will need it's own configuration profile. This is easily done using the stsTrunkBuilder.
It can create all the individual SipToSis configurations for you as well as
make scripts for starting and stopping the trunk. You can set it up on the same box as your PBX or separate.
See Skype Trunk Setup Guide for setup instructions.
What SIP PBX's will it work with?
It is known to work with plain Asterisk, AsteriskWin32, PBX-In-A-Flash, Elastix, SwitchVox, Axon, trixbox and 3CX.
It should work with almost any SIP PBX.
How many concurrent active Skype calls/channels can I make using this SIP Skype gateway with a PBX such as Asterisk?
You are limited by your available resources (memory, bandwidth and cpu). Here's the approximate per channel memory consumption and cpu usage for both supported multi channel platforms.
Windows - 56-70mb - Approx 12% cpu with an AMD Athlon 2700
Linux - 52mb+ for SipToSis & Skype and 16mb+ for Xvfb - Approx 5% cpu with an AMD Athlon 2700 using snd-dummy. If you use a different method to create X sessions, your memory usage will be different.
Conservative Ram Calculation: (#channels x 100MB) + OS ram requirements MB + Other APP requirements MB = TotalRam MB
CPU usage above is with standard PCMU codec (30ms packets) and RFC2833 DTMF.
You will have to experiment to find the limit for your environment.
You can always expand your channel count by adding more boxes. There are no built-in limits.
Use of a broadband internet service provider with sufficient bandwidth is required for multi channel operation.
I have run 10 channels myself - I do not have enough bandwith to run more. Some have run 40 channels.
If you run 40+ channels I recommend a dedicated 64 bit quad core server with 8gb ram running linux.
If you plan on using Skype's Subscription plans, please respect their Fair use policies. Using multiple subscription plans
in a multi channel setup will likely violate their policy. Please contact Skype directly regarding this.
What Skype client versions does this SIP to Skype Gateway work with?
For Sip to Skype on windows, at least 3.x. I recommend 3.6-3.8+ and 4.0.0.215 or higher. Callback and conference were tested with 4.1.0.141 version.
4.1.x version will not leave a voicemail message due to a Skype bug.
4.2.0.141 works with voicemail although Skype may hang or crash if you cancel during the greeting.
For Sip to Skype on Linux, it is known to work with 2.0.0.72 and 2.1.0.81.
Two stage callback with 2.1.0.47 with Skype as originator using Skype outbound does not work.
2.0.0.72 and 2.1.0.81 have not been tested with this feature.
Conference calling with 2.1.0.47 has audio problems (at least when using snd-dummy).
Again, 2.0.0.72 and 2.1.0.81 have not been tested with this feature.
Static versions are recommended to avoid memory usage problems.
For Sip to Skype on MAC OS X - 2.7.0.330 and 2.8.0.851.
It may work with other versions as well.
I want to use this with Asterisk/PBX, does it have to be on the same computer?
You can put it all on a single box or split components. The only restriction is that Skype
and SipToSis have to be together on the same computer.
For many channels, I would recommend that Skype/STS be on a separate computer
(especially windows) than your Asterisk box, but this is not a requirement.
How is the performance with multiple active calls?
Linux CPU usage under varying conditions.
Virtualbox (not an ideal setup) in 32 bit mode using a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup is a 2 card snd-dummy configuration. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
The Ubuntu distro is a 32 bit 9.0.4 (desktop) with 1.5GB total ram allocation. Approx 800MB was used for the entire setup.
10 Idle channels - There really are 10 they just are not "top" processes.
10 Active channels to echo test - Despite the apparent high load, there was no issue.
6 Active channels to echo test
Dedicated Ubuntu 64 bit 10.04 (desktop) on Dual Core AMD Athlon X2 4850e and 4GB ram.
Skype 2.1.0.81 using a single card snd-dummy configuration. SIP codec for all channels is PCMU.
10 calls were from another PC using softphones.
Top on 10.04 shows weird cpu % sometimes for idle processes.
10 Idle channels - There really are 10 they just are not "top" processes.
10 Active channels to echo test
6 Active channels to echo test
10 Active channels to echo test - This is using sndShare for sound device
Windows Vista CPU usage under varying conditions.
These shots are on a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup using on-board sound device. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
10 Idle channels
10 Active channels to echo test - This system maxed out at 8. It struggled to get 10 going.
6 Active channels to echo test
What should I know about the Callback and Conference features?
There are some compatability issues with certain Skype clients. Windows client 4.1.0.141 works well with both.
If making multiple PSTN calls via Skype, you will be charged for each outbound connection according to Skype's billing system.
In the case of a Skype subscription plan, each PSTN connection will use up time. Calls to Skype users are always free.
SIP side supports a single connection.
You can use a SkypeIn/Skype Online Number/SIP DID/Skype User/SIP User to initiate these calls.
You can combine a single SIP and multiple Skype targets (Skype or PSTN).
In the case of a DID, it must support callerID. Ask your provider directly if your country supports caller ID.
Callback is not legal in all countries, check it here.
Can you show 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.
Can you show 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.
Can you show 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.
What keeps unauthorized people from using my SIP or Skype accounts.
That is totally up to you. If you expose the SIP end to the internet, you can control access
by callerid, realm and IP Addresses/blocks combinations. You can also setup PIN authentication/dialing
by callerid, realm and IP Addresses/blocks combinations. For incoming Skype calls, you can control access by Skype User
or setup PIN authentication/dialing by Skype User.
How do I call a Skype User when I can only dial numbers on my phone?
SipToSis has powerful mappings and transforms to help with that. You can also translate numbers
to Skype users in your PBX or Skype client if it has speed dial support.
See the trouble shooting page for info.
Do I have to run the Skype client to use this Skype SIP Gateway?
Yes. This program does not bypass the Skype client.
All Skype Gateways operations are performed using the Skype API which means
SipToSis must be run on the same computer as Skype.
How do I uninstall SipToSis?
SipToSis does not use folders other than where you extracted it or create registry keys.
Simply delete the folder where you extracted the SipToSis program to "Uninstall" it.
Where is the source code?
For GNU GPL distributions, the source code is included in the distribution download.
This way you get the correct source for a particular distribution with no version confusion
and satifies the GNU GPL source availability requirement.
All this looks too complicated for me. Isn't there an easier way?
Click Skype Calls Without a Computer
to see devices that can use the Skype service without a computer or software.
I have seen this project called Sip-To-Sis, Sip-ToSis,Sip2Sis,SkypeToSis,Skype-To-Sis,Skype-ToSis,SkypeTo-Sis,SkyToSis,Sky-ToSis and SipTo-Sis. What is the official name?
The official name is SipToSis as shown on this site.
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