Speech synthesis for asterisk is an Asterisk AGI script that uses Google Translate to convert text to speech and play it back to the user. It supports a variety of different languages, local caching of voice data, and a choice between 8 kHz or 16 kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.
Category Archives: VOIP
Skype not working on T-Mobile USA IPv6 with UMTS u…
T-Mobile USA, Verizon LTE, and other are now supporting IPv6. I noticed that Skype does not work on the Android Galaxy Nexus with IPv6 on the T-Mobile USA network.
via Skype not working on T-Mobile USA IPv6 with UMTS u… – Skype Support Network.
RFC 3261 – SIP: Session Initiation Protocol
This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.
Free Local and Long Distance Calling with netTALK’s DUO
Free Local and Long Distance Calling with netTALK’s DUO.
North America $70/year, International $120/year. This seems like a Skype competitor. It looks like there’s a piece of hardware that comes with this that can be seen here.
Here‘s how it works. You don’t need a computer — just plug the device into the router and phone. I noticed an HTC device trying to connect SIP to this outfit so they must have an Android app as well.
Messaging apps, VoIP already eating into carrier revenues, study finds
The study, which was carried out on behalf of Mavenir by mobileSQUARED, found that a third of operators believe operator traffic from messaging, voice and video calling will decline between 11% and 20% over the next 5-10 years. Another 20% of operators expect even steeper declines in the 31% to 40% range.
via Messaging apps, VoIP already eating into carrier revenues, study finds.
IP Phone Systems | Zultys MX250
The MX250 IP PBX is a feature-rich enterprise-class business IP telephony system that supports high quality VoIP and Unified Communications services for up to 250 users on a single appliance and can be seamlessly expanded through an MXgroup network to support up to 10,000 users and 128 offices across an enterprise-wide Unified Communications system. The award-winning MX250 IP PBX is based on SIP Open Standards. In addition to Zultys’ enterprise-class IP phones, it can easily support legacy phones and a wide range of SIP-based third party devices to help companies more easily deploy the IP telephony system they need.
Speex: a free codec for free speech
Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNU Project and is available under the revised BSD license.
via Speex: a free codec for free speech.
Used for Siri.
Internet Real Time Lab (IRT)
The Internet Real-Time Lab (IRT) in the Computer Science Department at Columbia University conducts research in the areas of Internet and multimedia services: Internet telephony, wireless and mobile networks, streaming, quality of service, resource reservation, dynamic pricing for the Internet, network measurement and reliability, service location, network security, media on demand, content distribution networks, multicast networks and ubiquitous and context-aware computing and communication.
Phone Systems To Power Your Business
Seamless Communications at your Desk or on the Road
- A unified view of you from any phone.
- Tools that let your calls find you anytime, anywhere.
- Easy call transfers to and from all your phones
via Phone Systems To Power Your Business – Mobility – Digium® The Asterisk Company.