A telephony application very often needs to communicate information to the called party, and having multiple variations of recordings ready is notoriously difficult and results in a slow experience.
Our Text to Speech APIs offer a wide range of parameters to cater to every application without having to record audio files for the interactions.
A wide choice of languages and voices for each one makes it easy to build internationalized applications. SSML support provides the ability to speak dates, times, currencies, and many more formats without having to write extra code.
Communicating with your customers is as easy as typing out this sentence!
Voice applications are easier to use when you provide a way to interact that goes beyond simple DTMF. Speech recognition can be used for callers to select an option, to do language processing, or even keyword extraction.
The SignalWire speech recognition API enables your application to provide natural language interactions in a sophisticated way. Parameters include hints for better accuracy, various timeout controls, and the ability to do mixed mode input with DTMF.
With SignalWire speech recognition, it’s easy to keep track of your calls and interact with your users.
When placing phone calls, your application most likely needs to detect if the called party is a human or a machine. That way, you can customize your application’s response, whether it be leaving a message or hanging up.
SignalWire’s answering machine detection includes advanced proprietary technology that provides great accuracy and speed of detection.
Our intelligent APIs save time on each call by avoiding voicemails, fax machines, and disconnected lines, so you can reduce Cost Per Acquisition.
Conversations over the phone with your customers are an invaluable source of information that businesses should keep track of to analyze and make decisions. Recording calls can be useful in a variety of scenarios, from simple voicemail, to call quality monitoring, to performing natural language processing.
Recording live calls can help document these calls that may be reviewed later. SignalWire Recording API allows you to record participants from a call separately or in the same channel, and a rich set of webhooks is available to manage the recording lifecycle. Recording is asynchronous, meaning you can start a call recording at any time during a call, allow multiple call recordings at the same time, pause active call recordings to fill with silence or skip the pause, and record inbound calls without needing to join a conference or dial a number.
You can set recording to true whenever you use the API to make an outbound call, or you can use LaML’s <Record> in order to start recording within LaML Bins or your webhooks.
Often sensitive customer information shared over the phone during vital financial transactions is not stored properly. SignalWire’s Communication API allows for powerful call control capabilities for companies that are looking for ways to take down sensitive information without added liability and becoming PCI compliant. It lets a representative easily pause and resume the recording to take down sensitive information.
You can access the finished recordings within your SignalWire space, retrieve them using our REST API, or use recording status callbacks to send the completed recordings to your own server or database for storage.
All of the recordings are made available in WAV and MP3 format (WAV by default).
Recording a phone call is essential in keeping a record of all the interactions with the customers. Transcribing these recorded calls into plain text is also a powerful tool for businesses. Transcription of calls can be useful in ensuring compliance, training employees, analyzing feedback, and finding ways to optimize the customer experience.
Having the call contents in text format, compared to audio recording, is especially useful in search capabilities that help to analyze for patterns, etc. The use for this feature can range from voicemail to advanced language analysis.
Our transcription API provides a simple way to transcribe your recordings, through an asynchronous REST request that will be triggered when the transcription is ready, making your application fast and responsive.
You can easily set the parameter of <Record> called transcribe to true if you want your recordings to be transcribed. You can use the transcription status callback in order to have your transcripts sent to your own server, use the REST API in order to retrieve the transcription, or you can view them in your SignalWire space. Using our other APIs (or even utilizing our API with Zapier Webhooks and the existing SignalWire Zapier integration), you could easily set up a script where your completed recordings and transcriptions are sent via SMS (or email if you use Zapier) to the appropriate person/database.
Good customer experience is imperative for all businesses and providing a professional service is a big part of it. There are times when customers might be in noisy environments while on the call making it difficult to understand what they are saying. To understand their speech clearly to help them efficiently, it is useful to cancel the background noise.
SignalWire’s proprietary noise cancellation technology effectively improves the accuracy and quality of speech detection, without introducing any lag or performance degradation. It suppresses the background noise and provides the best audio experience. It is also easy to use through a single API command.
Conferences are a core component of many voice applications, from simple group calls to call center agents.
SignalWire’s conferencing API provides full control of participants, allowing you to moderate, whisper, and offer real-time coaching, all through simple LAML verbs.
A call whisper allows the callee (receiver of call) to receive an audio message before the call is connected and allows the callee to accept or reject the incoming call. The audio message can contain information such as the source or purpose of the call. During the whisper time, the calling party will be hearing a ringing tone until the callee accepts (or rejects) the call in progress.
You can also use the coach attribute on the conference in order to let a coach listen to a call and whisper to an agent in real-time to provide training or support.
Other helpful conference features include music designated to play during hold periods or before all the participants have joined, recording that can begin during or at the beginning of the conference, and alerts when a user enters or exits the conference.
The conference API allows you to search, modify, and manage conferences in your SignalWire account. A full set of REST API endpoints allows your application to query the conference state and return data about ongoing conferences, list all members and return specific information about them, manage recordings, update the in-progress conference or change something for a particular member, and more.
SignalWire’s LaML API supports the standard in VoIP telephony, the SIP protocol. This allows you to connect your devices and systems to the SignalWire Cloud, and to leverage the available features from the existing SIP infrastructure.
Using SIP with your SignalWire accounts allows you to build complex and useful systems without having to run your own infrastructure. You can make outbound calls via the API or redirect inbound calls to the SIP endpoint or SIP trunk that you have set up. If you are using SignalWire BYOC (Bring Your Own Carrier) in order to use your own carrier for SIP traffic, you can easily create a domain application (in your SignalWire Space under the SIP tab) to where you can point all of your normal SIP calls.
Read more about getting started with SIP here.
Build applications that include SIP endpoints. Our APIs let you place and manage SIP calls in addition to regular PSTN calls. Our Domain Apps feature enables you to receive SIP traffic at a hosted domain and route calls to your applications using SignalWire APIs.
Integrate SignalWire SIP endpoints for use with existing VoIP client applications, PBX, or call center systems. Route incoming PSTN calls to SIP endpoints and initiate outbound PSTN calls from SIP endpoints.
Our standard SIP trunking support enables connectivity to other IP telephony systems. Through our SignalWire Community Slack workspace, we have captured the collective experience from our customers utilizing a variety of different open-source and commercial IP telephony systems.
Transfer calls over SIP from your existing provider into our cloud infrastructure to provide you with the APIs and programmability to support modern applications. We include SRTP and TLS encryption at no extra cost.
Porting numbers from US carriers requires a Letter of Authorization (LOA) and getting the Customer Service Record (CSR). The process can take a couple of weeks depending how quickly the carrier responds. Porting of international phone numbers varies by country and relevant legislation. We will work with you in all the cases.