Source: andyabramson.blogs.com
Aaron Huslage has penned a nice introduction on Ostatic about FreeSwitch, a softswitch based on open source and open standard technology.
Some of my agency’s clients use it and love it, so I’m familiar with it more than most. It has a lot under the hood and is clearly disruptive in my view.
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project’s development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
OpenSER History
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Source: snapvoip.blogspot.com
FreeSWITCH is an open source telephony application built from the ground up
and designed to take advantage of most existing voip, telephony software and libraries. FreeSWITCH paves the way for one to build an open source PBX system, an open source voip switching platform or a VOIP, Telephony gateway uniting various technologies and platforms such as SIP H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle and OpenPBX, Bayonne, YATE or Asterisk.
FreeSwitch is a multi-platforms software and it runs on Windows Linux and MAC. It is also possible to run it on other UNIX flavors.
FreeSwitch is built on existing OpenSource libraries and projects, namely SQlite, SRTP Secure RTP, Apache Portable Runtime, Lib Resample, exosip , IAXclient, Speex Codec, libsndfile.
FreeSWITCH, the telephone soft-switch, has touched upon a few milestones combining many a famous VOIP application features into it’s core.
In Early April FreeSWITCH announced it’s interoperability capabilities with the GoogleTalk, goggle’s Voice chat program. Making it possible to gateway calls to SIP or the PSTN from Googletalk. Then again in July it brought out the capability of voice switching at 16Khz against the traditional 8Khx switching done in the VOIP world. This is a significant improvement on the quality of voice. What does that brings to a phone conversation? It brings more richness and clarity to voices, improving the overall experience of a phone call.
Now FreeSWITCH has done it again, Now it has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.
You want a twist with that? Yes you can have all that and more, interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.
Imagine, you start your VOIP/TELEPHONY/CHAT program, and a voice asking you for your account information, then in a chat window you type your name and another person on the other end, on a phone be able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether. And I am sure the idea will run wild through the VOIP community. Who knows, one slime might even try to patent the idea!
Source: snapvoip.blogspot.com
FreeSWITCH is an open source telephony application built from the ground up
and designed to take advantage of most existing voip, telephony software and libraries. FreeSWITCH paves the way for one to build an open source PBX system, an open source voip switching platform or a VOIP, Telephony gateway uniting various technologies and platforms such as SIP H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle and OpenPBX, Bayonne, YATE or Asterisk.
FreeSwitch is a multi-platforms software and it runs on Windows Linux and MAC. It is also possible to run it on other UNIX flavors.
FreeSwitch is built on existing OpenSource libraries and projects, namely SQlite, SRTP Secure RTP, Apache Portable Runtime, Lib Resample, exosip , IAXclient, Speex Codec, libsndfile.
FreeSWITCH, the telephone soft-switch, has touched upon a few milestones combining many a famous VOIP application features into it’s core.
In Early April FreeSWITCH announced it’s interoperability capabilities with the GoogleTalk, goggle’s Voice chat program. Making it possible to gateway calls to SIP or the PSTN from Googletalk. Then again in July it brought out the capability of voice switching at 16Khz against the traditional 8Khx switching done in the VOIP world. This is a significant improvement on the quality of voice. What does that brings to a phone conversation? It brings more richness and clarity to voices, improving the overall experience of a phone call.
Now FreeSWITCH has done it again, Now it has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.
You want a twist with that? Yes you can have all that and more, interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.
Imagine, you start your VOIP/TELEPHONY/CHAT program, and a voice asking you for your account information, then in a chat window you type your name and another person on the other end, on a phone be able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether. And I am sure the idea will run wild through the VOIP community. Who knows, one slime might even try to patent the idea!
Source: snapvoip.blogspot.com
FreeSWITCH is an open source telephony application built from the ground up
and designed to take advantage of most existing voip, telephony software and libraries. FreeSWITCH paves the way for one to build an open source PBX system, an open source voip switching platform or a VOIP, Telephony gateway uniting various technologies and platforms such as SIP H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle and OpenPBX, Bayonne, YATE or Asterisk.
FreeSwitch is a multi-platforms software and it runs on Windows Linux and MAC. It is also possible to run it on other UNIX flavors.
FreeSwitch is built on existing OpenSource libraries and projects, namely SQlite, SRTP Secure RTP, Apache Portable Runtime, Lib Resample, exosip , IAXclient, Speex Codec, libsndfile.
FreeSWITCH, the telephone soft-switch, has touched upon a few milestones combining many a famous VOIP application features into it’s core.
In Early April FreeSWITCH announced it’s interoperability capabilities with the GoogleTalk, goggle’s Voice chat program. Making it possible to gateway calls to SIP or the PSTN from Googletalk. Then again in July it brought out the capability of voice switching at 16Khz against the traditional 8Khx switching done in the VOIP world. This is a significant improvement on the quality of voice. What does that brings to a phone conversation? It brings more richness and clarity to voices, improving the overall experience of a phone call.
Now FreeSWITCH has done it again, Now it has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.
You want a twist with that? Yes you can have all that and more, interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.
Imagine, you start your VOIP/TELEPHONY/CHAT program, and a voice asking you for your account information, then in a chat window you type your name and another person on the other end, on a phone be able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether. And I am sure the idea will run wild through the VOIP community. Who knows, one slime might even try to patent the idea!
Source: snapvoip.blogspot.com
FreeSWITCH is an open source telephony application built from the ground up
and designed to take advantage of most existing voip, telephony software and libraries. FreeSWITCH paves the way for one to build an open source PBX system, an open source voip switching platform or a VOIP, Telephony gateway uniting various technologies and platforms such as SIP H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle and OpenPBX, Bayonne, YATE or Asterisk.
FreeSwitch is a multi-platforms software and it runs on Windows Linux and MAC. It is also possible to run it on other UNIX flavors.
FreeSwitch is built on existing OpenSource libraries and projects, namely SQlite, SRTP Secure RTP, Apache Portable Runtime, Lib Resample, exosip , IAXclient, Speex Codec, libsndfile.
FreeSWITCH, the telephone soft-switch, has touched upon a few milestones combining many a famous VOIP application features into it’s core.
In Early April FreeSWITCH announced it’s interoperability capabilities with the GoogleTalk, goggle’s Voice chat program. Making it possible to gateway calls to SIP or the PSTN from Googletalk. Then again in July it brought out the capability of voice switching at 16Khz against the traditional 8Khx switching done in the VOIP world. This is a significant improvement on the quality of voice. What does that brings to a phone conversation? It brings more richness and clarity to voices, improving the overall experience of a phone call.
Now FreeSWITCH has done it again, Now it has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.
You want a twist with that? Yes you can have all that and more, interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.
Imagine, you start your VOIP/TELEPHONY/CHAT program, and a voice asking you for your account information, then in a chat window you type your name and another person on the other end, on a phone be able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether. And I am sure the idea will run wild through the VOIP community. Who knows, one slime might even try to patent the idea!
