Tuesday, August 29, 2006
Skype Alliances
Skype and Intell
Skype* Internet calling starter kit: make free Skype to Skype calls, plus free calls within the US and Canada to mobiles and landlines until 12/31/06. For details, visit www.skype.com.
Skype VoIP demo
Hear the difference: Realize your competitive advantage, lower communication costs, and increase productivity with Skype* and Intel® dual-core technology.
Play the Video (12Mb)
(link)
Skype and Google Talk
Talking with Skype
Google Talk's got a new gig. Google and eBay have signed an agreement around text-based advertising and "click-to-call" advertising, in which Google Talk and Skype will power voice calls between customers and merchants. (Read the full press release here.)
Just as exciting are our plans to explore interoperability between Google Talk and Skype, making it easier for our users to chat with one another. This is just another step in our commitment to interoperability via open, industry standards.
Lewis Lin
Product Marketing Manager, Google Talk
(link)
Linphone (GNU oSIP, OpenSource)
Linphone is a web phone: it let you phone to your friends anywhere in the whole world, freely, simply by using the internet. The cost of the phone call is the cost that you spend connected to the internet.
Here are the main features of linphone:
- Works with the Gnome Desktop under linux, (maybe some others Unixes, but this has never been tested). Nevertheless you can use linphone under KDE, of course !
- Since version 0.9.0, linphone can be compiled and used without gnome, in console mode, by using the program called "linphonec"
- Works as simply as a cellular phone. Two buttons, and one more to chat.
- Linphones includes a large variety of codecs (G711-ulaw, G711-alaw, LPC10-15, GSM, SPEEX and iLBC ). Thanks to the Speex codec it is able to provide high quality talks even with slow internet connections, like 28k modems.
- Understands the SIP protocol. SIP is a standardised protocol from the IETF (http://www.ietf.org), that is the organisation that made most of the protocols used in the internet. This guaranties compatibility with most SIP - compatible web phones.
- You just require a soundcard to use linphone.
- Other technical functionnalities include DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses).
- Linphone is free software, released under the General Public Licence.
- Linphone is documented: there is a complete user manual readable from the application that explains you all you need to know.
- Linphone includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.
How to use linphone ?
To call somebody, you must provide to linphone a SIP URL:
It is something like toto@machine.com, where toto is a linux user that runs linphone, and machine.com is the name of a host on a network. If you don't know the machine's name you can specify simply an IP address in dot notation (as 192.0.0.1)
If you want more information about linphone, read first the user manual.
For documentation on the internals of linphone, see section « developers » of this web site.
Compatibility:
Linphone is mostly sip compliant. It works successfully with these implementations:
eStara softphone (commercial software for windows)
Pingtel phones (with DNS enabled and VLAN QOS support disabled).
Hotsip, a free of charge phone for Windows.
Vocal, an open source SIP stack from Vovida that includes a SIP proxy that works with linphone since version 0.7.1.
Siproxd is a free sip proxy being developped by Thomas Ries because he would like to have linphone working behind his firewall. Siproxd is simple to setup and works perfectly with linphone.
Partysip aims at being a generic and fully functionnal SIP proxy. Visit the web page for more details on its functionalities.
Linphone may work also with other sip phones, but this has not been tested yet. So if you want to complete the list by testing with your own sip phone, contact me.
Linphone uses the SIP protocol to establish calls, for that reason it cannot work with H323 phones, because SIP and H323 are different and opposite protocols. H323 phones are Netmeeting (for windows), Gnome-meeting (Unix), OpenPhone...
Monday, August 28, 2006
Skype Started Wifi program
Skype is a little piece of software that lets you make free calls over the Internet. We are revolutionising the telecoms world by allowing our users to call other Skype users anywhere in the world for free. In less than 2 years we’ve had over 180 million downloads (they’re free too) and we have a growing community of 55 million registered users.
Making free calls over Wifi is pretty nifty and our Wifi partners who have already opened their networks to Skype users agree. Making your network Skype-friendly means providing our users with free Skype-only web access. It’s a great way for new users to try Wifi and gives them an easy path to upgrade to full Internet access.
Here’s how to do it.
Marketing
Your Wifi start page should include a Skype banner, which you can get here.
You can check Skype’s promotional and distributional terms here.
There should also be a prominent space on your start page that lets people know they can use Skype for free from your hotspot. If you want to limit the time so that users sign-up for full Internet access, that’s fine, but we do request a minimum of 15 minutes free Skype-only access. We recommend “Talk for free! Click here to get free access to Skype.” Clicking on this should lead to a joint landing page.
The landing page should have a Skype banner, a link to download Skype and a button to allow Skype-users to connect to your Wifi network. Simple really.
Check out Skype’s affiliate program here as well. Using any of the banners provided in the affiliate program will allow you to earn commissions on sales that result from your clicks.
Technical
In the broadest possible terms, we consider an ideal network configuration to be one that’s set up according to the rules shown here:
Ideal network configuration for Skype
1. Outgoing TCP connections should be allowed to remote ports 1024 and higher.
2. Outgoing TCP connections should be allowed to remote ports 80 and 443.
3. Outgoing UDP packets should be allowed to remote ports 1024 and higher.
4. For ideal performance, the NAT translation should be stateful, meaning that translations are remembered and reused for subsequent packets. The state must be kept for at least 30 seconds after the most recent translation. (Skype recommends that the translations be maintained for as long as an hour, if possible.)
Many peer-to-peer applications, including Skype, rely heavily on UDP packets to help maintain the best possible quality of connection among peers because UDP packets can be transmitted quickly and require very little overhead to manage. However, for UDP communications to work properly for Skype through NAT, the translation rules for UDP packets must be consistently handled, meaning that UDP packets set from one external network address and port number must be consistently translated to an internal network address and port number without varying either the network address or port number.
Although the use of UDP is optional - meaning Skype will work fine without the ability to transmit UDP messages - the call quality experienced by Skype users will be much better, on average, if the caller is able to send UDP packets to the called party and receive UDP answers in reply.
Tip: Checking your network for P2P friendliness
Many of our customers have told us that they use a freeware program called “NAT Check”, written by Bryan Ford, to see if their network’s UDP translation is compatible with P2P protocols including Skype. The NAT Check program is available for free download from the program’s website at http://midcom-p2p.sourceforge.net/ and is available in a precompiled form for platforms running Microsoft Windows, Mac OS X and Linux. (NAT Check is not Skype software.)
Skype-friendly partial results from NAT Check
UDP RESULTS:
In the results of NAT Check shown above, we see that the network’s UDP translation is applied consistently (“consistent translation”), that the input and output ports are identical except in the event of a conflict (“loopback translation”) and that unsolicited UDP packets sent to the network are discarded (“unsolicited messages filtered”).
Although not strictly necessary, it is preferable for the network’s firewall or NAT gateway to support IP packet fragmentation and reassembly. In addition, the firewall must not block an attempt to send parallel UDP packets or TCP connection attempts to multiple ports at the destination address. Some firewalls misclassify such behavior as port scanning and therefore block the host altogether. Such behavior could not only impact the ability of Skype to run but would likely impact other legitimate network applications running on the same host computer.
by Skype.com
Skype is a little piece of software that lets you make free calls over the Internet. We are revolutionising the telecoms world by allowing our users to call other Skype users anywhere in the world for free. In less than 2 years we’ve had over 180 million downloads (they’re free too) and we have a growing community of 55 million registered users.
Making free calls over Wifi is pretty nifty and our Wifi partners who have already opened their networks to Skype users agree. Making your network Skype-friendly means providing our users with free Skype-only web access. It’s a great way for new users to try Wifi and gives them an easy path to upgrade to full Internet access.
Here’s how to do it.
Marketing
Your Wifi start page should include a Skype banner, which you can get here.
You can check Skype’s promotional and distributional terms here.
There should also be a prominent space on your start page that lets people know they can use Skype for free from your hotspot. If you want to limit the time so that users sign-up for full Internet access, that’s fine, but we do request a minimum of 15 minutes free Skype-only access. We recommend “Talk for free! Click here to get free access to Skype.” Clicking on this should lead to a joint landing page.
The landing page should have a Skype banner, a link to download Skype and a button to allow Skype-users to connect to your Wifi network. Simple really.
Check out Skype’s affiliate program here as well. Using any of the banners provided in the affiliate program will allow you to earn commissions on sales that result from your clicks.
Technical
In the broadest possible terms, we consider an ideal network configuration to be one that’s set up according to the rules shown here:
Ideal network configuration for Skype
1. Outgoing TCP connections should be allowed to remote ports 1024 and higher.
2. Outgoing TCP connections should be allowed to remote ports 80 and 443.
3. Outgoing UDP packets should be allowed to remote ports 1024 and higher.
4. For ideal performance, the NAT translation should be stateful, meaning that translations are remembered and reused for subsequent packets. The state must be kept for at least 30 seconds after the most recent translation. (Skype recommends that the translations be maintained for as long as an hour, if possible.)
Many peer-to-peer applications, including Skype, rely heavily on UDP packets to help maintain the best possible quality of connection among peers because UDP packets can be transmitted quickly and require very little overhead to manage. However, for UDP communications to work properly for Skype through NAT, the translation rules for UDP packets must be consistently handled, meaning that UDP packets set from one external network address and port number must be consistently translated to an internal network address and port number without varying either the network address or port number.
Although the use of UDP is optional - meaning Skype will work fine without the ability to transmit UDP messages - the call quality experienced by Skype users will be much better, on average, if the caller is able to send UDP packets to the called party and receive UDP answers in reply.
Tip: Checking your network for P2P friendliness
Many of our customers have told us that they use a freeware program called “NAT Check”, written by Bryan Ford, to see if their network’s UDP translation is compatible with P2P protocols including Skype. The NAT Check program is available for free download from the program’s website at http://midcom-p2p.sourceforge.net/ and is available in a precompiled form for platforms running Microsoft Windows, Mac OS X and Linux. (NAT Check is not Skype software.)
Skype-friendly partial results from NAT Check
UDP RESULTS:
- UDP consistent translation: YES (GOOD for peer-to-peer)
- UDP loopback translation: YES (GOOD for peer-to-peer)
- UDP unsolicited messages filtered: YES (GOOD for security)
In the results of NAT Check shown above, we see that the network’s UDP translation is applied consistently (“consistent translation”), that the input and output ports are identical except in the event of a conflict (“loopback translation”) and that unsolicited UDP packets sent to the network are discarded (“unsolicited messages filtered”).
Although not strictly necessary, it is preferable for the network’s firewall or NAT gateway to support IP packet fragmentation and reassembly. In addition, the firewall must not block an attempt to send parallel UDP packets or TCP connection attempts to multiple ports at the destination address. Some firewalls misclassify such behavior as port scanning and therefore block the host altogether. Such behavior could not only impact the ability of Skype to run but would likely impact other legitimate network applications running on the same host computer.
by Skype.com
SIP Over Asterisk
How do I get SIP endpoint to talk through Asterisk?
I have two SIP endpoints provisioned on the SIP PBX and I can see SIP registers correctly. However, the two SIP endpoints can not call one another (get a "404 Not Found" SIP error message in response to an INVITE). I want to add more SIP endpoints to the PBX but need to get this simple case working first.
I have configured the sip.conf file as follows:
1000
type=user
host=dynamic
context=default
username=1000
dtmfmode=rfc2833
1001
type=user
host=dynamic
context=default
username=1001
dtmfmode=rfc2833
The sip clients are registering in the format:
Any ideas on what is wrong? Do I need to do anything with the extensions.conf file (or is the default settings sufficient to get one sip phone to call the other?)
by voip-info.org
