The Direct Routing feature in Microsoft Teams has become the preferred method of connecting users to the PSTN, with media reports indicating upwards of 80% of organizations choosing this option over Calling Plans. Now, Twilio has joined forces with Ribbon Communications to provide enterprise IT managers with a new PSTN connectivity solution for Microsoft Teams that offers truly global coverage, low calling rates and on-demand provisioning.
The joint solution combines Twilio Elastic SIP Trunking with Ribbon SBCs to connect Teams users to the PSTN in up to 100 countries. It enables you to deploy Microsoft Teams with Direct Routing on a global basis with the simplicity of a single, centrally managed solution.
Direct Routing vs Microsoft Calling Plans
IT managers who are rolling out Microsoft Teams for their unified communications service have a choice between two PSTN connectivity options: Direct Routing connects Teams users to a third party SIP trunking …
Considering the rapid growth of the Microsoft Teams UCaaS service, it’s no surprise Twilio is often asked to provide a PSTN connectivity solution for the service. Enterprise customers want to combine the rich telephony features offered by Microsoft Teams with the global coverage and economical calling rates offered by Twilio SIP trunking services. And now – they can!
We’re excited to collaborate with AudioCodes to introduce the direct routing connectivity of our Elastic SIP Trunking service to Microsoft Teams. The combination enables any organization using Microsoft Teams to connect to the PSTN via Twilio Elastic SIP Trunking. Additionally, users can get coverage in up to 100 countries with on-demand provisioning, unmatched reliability, and low calling rates.
“We are delighted to be collaborating with Twilio to deliver scalable Microsoft Teams Direct Routing connectivity,” said Nimrode Borovsky, VP and General Manager, Enterprise, at AudioCodes. “As Microsoft Teams gains in popularity, not …
SIPレジストレーションを行うことで、SIP対応デバイスやソフトフォンをTwilioに直接接続することができます。またProgrammable Voice APIを使用して、複雑なハードウェアやネットワークへの物理接続を必要とせずに、強力なコール処理ロジックを構築することができます。おなじみのTwilioアプリケーションを使って、PSTNやWebRTC/モバイルクライアントと同じようにSIPエンドポイントに接続することができるのです。しかしこれまでは、レジストレーション済みSIPデバイスはグローバルにコールを発着信することができるものの、SIPレジストレーション先は、米国バージニア州アッシュバーン(米国東部)にある当社データセンターに限定されていました。
SIP Registration allows you to connect your SIP-enabled devices or softphones directly to Twilio, and use the Programmable Voice API to build powerful call-handling logic without standing up complex hardware or physical connections to your network. You can use the Twilio applications you know and love to connect to SIP endpoints the same way you do to the PSTN or WebRTC/Mobile clients. Until now, however, while your registered SIP devices could make and receive calls globally, your endpoints could only register to Twilio via our US-East data center in Ashburn, VA, USA.
Earlier this year, we announced new Twilio Edge Locations to improve application performance. With this launch, we are building on top of what we delivered to improve the end-user experience.
Today, we are happy to announce that Twilio SIP Registration is now accessible via our public and private Edge Locations worldwide. Now you can register your SIP device …
As we continue to grow and develop our Programmable Voice portfolio, we are excited to announce the addition of new capabilities to enhance the overall user experience. These features will enable you to use the applications you have developed for PSTN and Client access interchangeably with your SIP endpoints too.
Support for Blind Transfer with addition of inbound SIP REFER to Twilio
The SIP REFER method enables moving an active SIP session from one SIP endpoint to another. In other words, REFER is used to implement call transfers on your SIP-connected calls. Until now, Twilio has only supported sending outbound SIP REFER messages to your SIP network, using the TwiML
Today, we are announcing support of inbound SIP REFER messages, meaning you now can send SIP REFER messages from your SIP network to Twilio in order to initiate call handling in your application. Effectively, this means that Twilio …
Twenty years ago, when SIP trunking and VoIP were introduced on the world stage, we hadn’t yet dreamed of all of the ways that the internet would revolutionize how we connect with each other. Now, using the platform of our preference, unified communication online has become a part of our everyday lives.
Phone calls still hold strong as a tried-and-true method of communication in a world that loves text, email, and messaging. Increasingly, businesses are leaning on VoIP and SIP trunking to meet the demand for affordable (and scalable) person-to-person communication.
As we increase our reliance on online communication, SIP trunking frequently becomes the subject of conversation regarding digital transitions. Although SIP trunking is a hot topic, it can be easy to gloss over the basics of how it works and why it can be transformative for your business. Never fear, we’ve got you covered! Here’s everything you need …
The world has changed, thanks to the Internet. We are able to collaborate and build quickly and efficiently. But – this phenomenon has also increased the number of security risks and attempts by hackers to extract valuable information.
Here at Twilio we use Rest APIs which are secure by default. On top of that, our Elastic SIP Trunks can be secured by simply flipping a switch.
If you have not turned on the secure feature for Elastic SIP Trunk, this post will give you the reasons you should do it. We’ll show you how to enable Secure Trunking and walk through configuring a Cisco Unified Border Element (CUBE) with TLS & SRTP.
TLS/SRTP is now included 🎉
We are now offering this security feature free – all you need to do is flip the switch! Learn about the TLS specification for secure SIP trunking in our docs.
We have already …
Contact center managers and architects planning a migration to the Genesys Cloud service will encounter an important decision that they may not have given a lot of thought to. It’s about how best to connect their customers to Genesys Cloud. There are multiple PSTN connectivity options available to Genesys customers and each presents trade-offs that affect business agility, cost, customer experience and other factors.
Genesys offers a “bring your own carrier” (BYOC) approach, which gives organizations the flexibility to select their voice services for Genesys Cloud. This gives Genesys customers access to a range of PSTN connectivity services to fit their business needs. You can leverage global SIP trunking services from third-party providers, including Twilio, and realize geographic coverage, pricing, flexibility and other advantages.
Native Genesys Voice Services Offer Solutions for Any Deployment
The native Genesys Cloud voice services offer you the simplicity of one-stop shopping. It’s incredibly easy to …
Selecting a SIP trunking service provider may seem like a straight-forward decision to many IT managers. After a brief evaluation, they might conclude the services are commodities with little differentiation, especially given the fact SIP trunking has been available for more than a decade and adoption is widespread in most industrialized countries. But reality is quite different, there are several factors that differentiate them.
The SIP trunking services market is vibrant and dynamic with many providers competing for business with innovative offerings. Two broad provider categories have been established: traditional wireline telecommunications providers and communications platforms as a service. The traditional companies are often referred to as telcos, while the communication platform companies are referred to as CPaaS or cloud companies. These categories are differentiated by their investments in wireline transmission facilities.
Depending upon an organization’s needs, a telecommunications provider may be a better fit than a CPaaS, or vice-versa. …
In these unprecedented times of social distancing and a virus pandemic, we are seeing a huge increase in the number of voice calls being made. Our Super Network team is working around the clock to ensure that calls make it through. However, in some areas, we are seeing that lines from carriers are so congested that the calls are never reaching Twilio. Here in this tutorial, we will outline some options to bypass the congestion on the PSTN by using Twilio Client.
Twilio Client is a WebRTC based voice client that works in browsers and mobile devices using a data connection. Once a client call is connected to Twilio it can be routed directly to your existing infrastructure via SIP or uncongested PSTN paths allowing calls to continue to flow.
Currently, your customers and staff may be struggling to connect to your contact centers or business-critical services …