Yealink, the global leading unified communication (UC) terminal solution provider, announced today the completion of Yealink latest phone array T4S Series (including T48S, T46S, T42S and T41S) and T27G interoperability testing with FreeSWITCH, a scalable open source cross-platform telephony platform (FS version 1.6~64bit). As the world’s first cross-platform scalable free multi-protocol soft switch, FreeSWITCH is designed to route and interconnect popular communication protocols using audio, video, text, or any other form of media. With today's release of the latest compatible Yealink phones, businesses employing a FreeSWITCH-powered VoIP infrastructure can rely on Yealink phones to benefit from the robust and cost-effective unified communications (UC) experience.
"Interoperability between software and hard phone devices is vital in connecting the user to the full communication experience. We at FreeSWITCH are delighted to have the opportunity to work with manufactures to ensure the greatest quality application for the end users and a full FreeSWITCH feature experience." said Brian West, QA and Development Director at FreeSWITCH Solutions.
“We are very excited to see the successful interoperability tests with FreeSWITCH,” said Yealink Vice President Stone Lu. “By working together, FreeSWITCH customers can now access advanced telephony features with Yealink phones that help them improve their communication efficiency and productivity,” he added. “We very much look forward to future cooperation with FreeSWITCH as we continue delivering more innovative endpoint solutions in the global telecommunication market space.”
The compatible Yealink T4S series is the upgraded IP phone line of Yealink’s former T4 series. Designed for today’s busy executives and managers, the T4S series features an elegant appearance, Optima high-definition audio quality and a remarkable phone experience. Integrating cutting-edge features like Wi-Fi and Bluetooth connectivity, the T4S series enriches business users’ daily collaboration. The SIP-T27G IP phone, as the upgraded product of T27P, is Yealink’s latest feature-rich tool unifying superior voice capabilities and increased function extension capability for business. For more Yealink phone information, visit Yealink’s official IP phone page.
First released in January 2006, FreeSWITCH has grown to become the world’s premier open source soft-switching platform. This versatile platform is used to power voice, video, and chat communications on devices ranging from single calls on the Raspberry Pi to large server clusters handling thousands of calls. FreeSWITCH powers a number of commercial products. Released under the business-friendly MPL 1.1 open source license, FreeSWITCH is continuing the open source telephony revolution that is occurring worldwide.
Yealink, the global leading unified communication (UC) terminal solution provider, helps businesses of all sizes make the most of their UC experience and embrace the power of “Easy Collaboration”. Yealink One-stop UC Terminal Solutions unify voice, video and data, and satisfy diverse customer needs and usage scenarios. The company’s comprehensive product portfolio includes video conferencing systems, conference phones, desk IP phones, wireless DECT phones and accessories. Customers from more than 100 countries enjoy Yealink’s reliable UC terminal solutions through its well-established global sales and service network. For more information, please visit: www.yealink.com.
The FreeSWITCH 1.6.17 release is here!
This is just a routine maintenance release, but there are some great features in this release as well.
Release files are located here:
- Tarball: http://files.freeswitch.org/releases/freeswitch/freeswitch-1.6.17.tar.gz
- Packaging: https://freeswitch.org/confluence/display/FREESWITCH/Installation
New features and improvements that were added:
- FS-10210 [mod_console] Added support for uuid config param and 'console uuid' api command to make reading logs from FreeSWITCH easier while running it in containers that read logs from stdout.
- FS-10126 [core] General Video Improvements
Improvements in build system, cross platform support, and packaging:
- FS-10153 [build] Fixed mod_http_cache build on FreeBSD
The following bugs were squashed:
- FS-10220 [mod_conference] Fixed an issue with conference channel parameters not working
- FS-10099 [mod_conference] Fixed a rare segfault on race condition when shutting down a conference
- FS-10225 [mod_conference] Fixed an off-by-one error on the calculation of the correct layout to use when choosing a layout group while a file is playing.
- FS-10059 [sofia-sip] Added a fix to correctly handle re-invites during t.38 call
- FS-9765 [mod_sofia] Fixed to use switch_channel_var_true instead of switch_channel_get_variable to prevent allocating on every hold/unhold just to check if this is enabled
- FS-10117 [mod_rayo] Fixed it to allow duplicate rayo signal-type configs for call progress detector to prevent segfaults
- FS-10100 [mod_av] Fixed a crash on allocation error and other error cases when opening a file
- FS-10209 [core][mod_av] Add auth_username and auth_password to media params for authentication when using streams
- FS-10150 [core] Reduce writes to closed ssl sockets
- FS-10195 [core] Fixed intermittent segfaults
- FS-10222 [core] Added disable_audio_jb_during_passthru and disable_video_jb_during_passthru default behavior leaves the jitter buffer on for audio and video respectively during passthru situations unless the new variables are set to true.
- FS-10233 [mod_local_stream] Fixed a segfault caused by trying to read a music file that is not open while playing a chime
The FreeSWITCH team was recently asked to speak to a IP Telephony class. This class is part of the masters in telecommunications systems management program within Northeastern's graduate school of engineering. The students are part of a program that provides a comprehensive overview of IP telephony architectures and protocols, with emphasis on SIP, the Session Initiation Protocol.
Topics covered in the class include a review of classical circuit-switched telephony, especially signaling; a review of IP networking, especially routing and addressing; peer and master-slave protocols for IP telephony (SIP, H.323, MGCP); speech coding; the transport of real-time traffic over IP (RTP and RTCP); bandwidth control; issues in network quality of service, such as traffic modeling, dimensioning, and QoS mechanisms; IMS; messaging protocols and systems; browser and API driven telephony applications, and Unified Communications. Emphasis on SIP includes call flows, network components, and services. Focus is placed on execution of hands on labs and semester long projects to provide hands on learning in a small team environment.
The FreeSWITCH team presented using the Verto communicator to discuss the ins and outs of FreeSWITCH as well as discussing some of the finer points of of the industry from a FreeSWITCH perspective. Starting with the origin of FreeSWITCH and then leading into the features and common configuration questions and giving these students the opportunity to ask questions and hear the origin story of a leading open-source communication project first-hand. The team was honored to be able to reach out to the next generation of communications enthusiasts. We hope you enjoy listening to this presentation as much as we enjoyed putting it together.