Voicegain is excited to announce the launch of Voicegain Casey, a payer focused AI Voice Agent that transforms the end-to-end call center experience with the power of generative AI. Voicegain Casey is a software suite of the following three Voice AI SaaS applications that helps a health plan or TPA call center improve operational efficiency and increase the CSAT and NPS (Net Promoter Score):
The AI Voice Assistant replaces a touch-tone IVR with a modern LLM-powered conversational AI Phone Agent. The AI Phone Agent can answer all calls that are received at a Health Plan or TPA Call center. It engages callers in a natural conversation and automates routine telephone calls like Claims Status, eligibility inquiries and eligibility verifications. In our experience, there is a very compelling business case to automate provider phone calls in Health Plan and TPA call centers and Voicegain Casey is specifically designed to do this. The AI Voice Assistant is also trained to perform HIPAA Validation and triaging of calls. So if the AI has not been trained to answer a specific question, it routes the call to the call center for live assistance.
Voicegain AI Co-Pilot is a browser extension that runs as a browser side-panel of Call Center Agent's CRM. The Co-Pilot is integrated with the Contact Center/CCaaS platform of the Payer. When a call transferred by the AI Voice Assistant is eventually answered by a Live Agent, all the information collected by the AI Voice Assistant is presented as a "Screen-Pop" on the Desktop of the Live Agent (also referred to as CTI). This CTI/Screen pop feature ensures that the front-line call center staff do not have to ask the customer to repeat any information that was provided to the AI Voice Assistant. In addition to the Screen-Pop, the AI Co-Pilot also guides the front-line call center staff in real-time by listening, transcribing and analyzing the conversation and providing real-time guidance . The AI Co-Pilot also generates a summary of the conversation within five seconds of the completion of the call. This automated summarization easily saves 1-2 mins of wrap-up time or after call work which is very common in these health plan and TPA call centers.
Voicegain AI QA & Coach is a browser-based AI SaaS application that is used by Team-leaders, QA Call Coaches/Analysts and Operations Managers in a call center. This AI SaaS app can record and measure the sentiment of the callers, analyze the QA score and provided automated coaching tips to the Agents. Voicegain uses the latest open-source reasoning LLMs (like LLAMA 3, Gemma) and closed-source reasoning models like o-3 from Open AI. With the power of modern reasoning models, almost the entire QA score-card (at least 80% of the questions) can be easily answered with modern reasoning-based LLM models. This SaaS App also provides a database of all whole-call-recordings of the entire conversation of the customer - which includes the AI Voice Assistant part, the transfer to the specific Call Center queue and eventually the entire conversation between the Live Agent and the Caller.
Voicegain Casey requires the following 3 key integrations to help with automation and real-time assistance.
Voicegain Casey integrates with modern CCaaS platforms. Current Integrations include Aircall, Five9, Genesys Cloud. Planned integrations include Ringcentral, NICE CXOne and Dialpad.
Voicegain Casey integrates with the CRM software of the Health plan or the TPA. This can be an off-the-shelf CRM like Zendesk or Saleforce. It can also be a proprietary/homegrown CRM. As long as the CRM is a browser-based SaaS application, this should not be an issue. Voicegain Casey AI Co-Pilot is a browser-extension that is installed in the side-panel of the same browser tab as the CRM. At the end of the call, the summary of the call is automatically generated and available on the browser extension within 5 seconds of the end of the call.
Voicegain Casey needs access to the member data (for HIPAA Validation) and claims data.
For further information on Voicegain casey, including a demo, please visit this link
If you would like to understand Voicegain Casey in more detail or if you would prefer a detailed product demo over a Zoom video call, please do not hesitate to send us an email. You can reach us at sales@voicegain.ai or support@voicegain.ai
FreeSWITCH is a very capable telephony platform suitable for building various telephony applications. Some of those applications will rely speech-to-text conversion, for example: ACDs (automatic call distribution), IVRs, Voice-Bots, Real-Time Agent Assist, real-time conference call transcription, call monitoring, etc.
Voicegain Speech-to-Text platform can be used with FreeSWITCH in a variety of ways.
Voicegain STT platform has supported MRCP (Media Resource Control Protocol) for a long time now. Our ASR can be accessed using MRCP and we support both grammar-based recognition (e.g. GRXML) and large-vocabulary transcription. MRCP is a communication protocol designed to connect telephony based IVRs and Voice Bots with speech recognizers (ASR) and speech synthesizers (TTS).
FreeSWITCH can interact with MRCP based recognizers using the included mod_unimrcp module. Voicegain STT has been tested with mod_unimrcp and interfaces with it without problems. You can learn more about using Voicegain STT via mod_unimrcp in this blog post.
Voicegain supports MRCP both in the Cloud and on the Edge (on-prem). We will soon be releasing in OpenSource a recognizer plugin for unimrcp server that will give you even more options in deploying FreeSWITCH with Voicegain and MRCP.
Voicegain provides a Telephony Bot API which is a callback API - similar in style to Twilio TwiML. You can place a call to Voicegain endpoint either using a phone number obtained from Voicegain or using a SIP endpoint unique to your Voicegain application. When a call arrives you will get a web callback and the response you will provide will determine actions that the Voicegain platform will perform, like e.g. play a prompt, recognize speech, detect DTMF, etc.
You can learn more about this API from the following blog posts:
If you have a FreeSWITCH application and you would like to recognize spoken speech you can bridge into Voicegain SIP endpoint and in a callback specify a prompt and the type of speech capture (grammar-based or large vocabulary). Once the recognition finishes you will get a callback and then you can either issue a disconnect command which will transfer call flow back to your Freeswitch app, or you can continue with additional questions and recognitions on Voicegain platform as needed.
Below is an example of a simple interaction with 4 participants:
This is still not Generally Available - please contact us if you are interested in testing.
mod_voicegain will give you capabilities similar to using mod_unimrcp with Voicegain but without the whole overhead of using an MRCP protocol - mod_voicegain talks directly to Voicegain ASR.
mod_voicegain taps into the FreeSWITCH inbound audio stream and sends the audio data to Voicegain ASR in the Cloud or on the Edge. Voicegain ASR processes the audio according to the invocation parameters specified in the data argument. It then communicates the result of transcription or recognition in an Event.
mod_voicegain installs on FreeSWITCH as an app and can be invoked as a such, e.g.:
or from LUA script:
Results will always be returned as a FreeSWITCH event but it is also possible to get the results in a callback to the url specified in callback.uri
The FreeSWITCH event will be of custom type (Event-Name: CUSTOM) and Event-Subclass will be "voicegain_asr_update". The relevant payload will be in the "ASR-Response" field formatted as JSON.
You can read more about mod_voicegain is this Knowledge Base article.
mod_vg_tap has been developed with applications like Real-Time Agent Assist in mind. These apps need access to the audio stream from a FreeSWITCH call but do not otherwise need to interact with FreeSWITCH (unlike IVR and Voice-Bots).
mod_vg_tap installs as an app and has simple commands to start/stop streaming to Voicegain Speech-to-Text engine.
The start command can specify the following destinations:
The results from transcription are generally not returned to a FreeSWITCH app but will be delivered to the destination specified when starting speech-to-text session - the results can be delivered via websocket, polling, or callback.
If you want more information about any of these methods of integrating Voicegain with FreeSWITCH, please email us at support@voicegain.ai.
Dallas, Texas - October 26, 2021: OnviSource, a leading provider of intelligent automation solutions for workforce optimization, contact center operation analytics and automation, customer experience management, and business process automation, announced today a strategic partnership with Voicegain, an innovative Speech-to-Text/ASR company. OnviSource has integrated Voicegain’s deep learning-based speech-to-text platform into its Intellecta™ multichannel analytics solution which utilizes speech-to-text and natural language understanding to analyze customer interactions and audio-based content to discover actionable knowledge and extract business insights.
OnviSource will leverage the Voicegain platform to serve its growing enterprise client base from various industries such as nationwide wireless service providers, banking, financial services, utilities, insurance and others.
“We are pleased to announce this partnership with Voicegain as their AI-driven ASR further augments our AI-driven intelligent automation solutions and our hyper-automation platform that offers integrated AI, conversational AI, RPA, BPA and analytics,” said Ray Naeini, Chairman and CEO of OnviSource. “Our partnership will allow both companies to jointly develop highly sophisticated and customized AI models for various applications and industries in order to deliver unmatched accuracy and performance.”
To achieve high performance, OnviSource deployed the Voicegain ASR Engine on servers with NVIDIA GPUs in its data center. This architecture is referred to as an Edge deployment. While Voicegain also offers a multi-tenant cloud solution, an Edge deployment architecture has two important benefits for OnviSource.
The first major benefit is that it allows OnviSource to meet strict customer contractual commitments related to data privacy, security and control. The second benefit is that it delivers approximately a 75% reduction in costs for OnviSource compared to usage-based pricing models provided by other providers, empowering OnviSource to offer its feature-rich solutions at highly affordable and flexible prices.
“We are excited to be selected by OnviSource for its call center and enterprise speech analytics products. This decision validates the ‘3As’ on which Voicegain differentiates itself in the ASR market – Accuracy, Affordability and Accessibility,” said Arun Santhebennur, Co-founder & CEO of Voicegain. “Our joint product enhancements will deliver highly accurate Speech-to-Text models for complex business applications.”
Selection of the Voicegain product by OnviSource was based on comprehensive trials and pilot programs related to accuracy, performance and applicability of Voicegain’s product, combined with detailed comparative analysis with other products in the market.
Additionally, the Voicegain product offers simplicity in deployment and usage as the entire platform is deployed on a Kubernetes cluster. Its Edge deployment offers a simple script to download and deploy all the packages and dependencies on any server with NVIDIA GPUs.
About OnviSource
For more than a decade, OnviSource has enabled several hundred small-to-large companies across a broad range of industries to cost-effectively manage, automate and improve their customer experience and business processes by offering advanced solutions in multichannel data and media capture, unification, analysis, decision making and automation for their entire enterprise, including their contact centers, back offices and IT organizations.
OnviSource ia.Enterprise Intelligently Automated (IA) solutions offer Workforce Optimization and Workforce Management (WFO/WFM), inclusive Teleservice Customer Engagement Management, Multichannel Customer Engagement Analytics, intelligently automated Customer Survey, Process Automation through Robotic Process Automation (RPA) and Intelligent Process Automation (IPA) and Intelligent Virtual Agent (IVA). The Company delivers its solutions as software products, cloud or Software-as-a-Service (SaaS), managed services, or any combination. OnviSource’s special Advantage Platinum program assures that solutions work for customers’ specific needs by offering a series of customer assistance programs with no obligations. These programs include consultation, proof-of-concept and hands-on operation assistance. OnviSource is headquartered in Plano, Texas (North Dallas area), with an additional operation center in Oklahoma.
About Voicegain
Voicegain is a deep neural network-based Speech-to-Text platform that is focused on developers of voice applications. Voicegain offers a full suite of APIs, SDKs and SaaS apps on top of its platform to automate and analyze voice-based interactions in contact centers, sales and meetings. To learn more, visit Voicegain.ai or create a free account to get started.
Press Contact:Voicegain: Arun Santhebennur, CEO
OnviSource: Deborah Cromwell, Marketing Manager
deborah.cromwell@onvisource.com
One of the previous blog posts described a Voice Bot built using Twilio, Voicegain, RASA, and AWS Lambda. Twilio was used for telephony (phone numbers, SIP Trunking, TwiML for call control) Voicegain provided the ASR/speech recognition, while AWS Lambda was coordinating the actions. The setup works but is involved. The need to pass the speech recognition results via S3 (as Lambda is stateless and does not have memory between function calls) may occasionally cause delays in requests and responses.
Voicegain now integrates with Amazon Chime Voice Connector to offer a pay as you go SIP Trunking service directly from the Voicegain web console. You can also purchase phone numbers and receive inbound calls. Support for making outbound Speech IVR calls is in the works.
Of course, we continue to support developer that use Twilio and SignalWire using simple SIP INVITE - this blog describes how.
The sequence diagram is provided below. It is very simple. Basically, the sequence of operations is as follows:
The sample code for the Lambda function (in python and node.js versions) is available on our github.
1. Click here for instructions to access our live demo site.
2. If you are building a cool voice app and you are looking to test our APIs, click hereto sign up for a developer account and receive $50 in free credits
Voicegain Speech-to-Text platform has already for a while supported many of the Twilio features like:
Release 1.26.0 of the Voicegain platform finally offers a full 2-channel support for Twilio Media Streams. This enables real-time transcription of both the inbound and outbound channels at the same time.
Twilio <Stream> command takes a websocket url parameter as a target to which the selected channels are streamed, for example:
The wss url can obtained by starting a new Voicegain real-time transcription session using https://api.voicegain.ai/v1/asr/transcribe/async API. The session part of the request may look like this (notice that two session are started and each will be fed different channel left/right of the audio stream):
We also need to tell Voicegain to take input in TWIML protocol in stereo:
Notice that we can enable audio capture which in addition will give us a stereo recording of the call once the session is complete.
In the response of the start of Voicegain session we get 3 websocket urls:
On our github we provide an example python code that starts a simple outbound Twilio phone call and then transcribes in real-time both inbound and outbound audio.
The sample code illustrates an outbound calling example which is somewhat simpler because there are no callback involved. In a case of an inbound call, the request to Voicegain would have to be done from your Twilio callback function that gets invoked when a new call comes in, otherwise, the rest of the code would be very similar to our github example.
Some of these are already listed on Twilio Media Streams page:
We will be testing the <Stream> functionality on the LaML command language provided by SignalWire platform which is very similar to Twilio TwiML - we will update our blog with the results of those test.
We are also working on a real-time version of our Speech Analytics API. Once complete then all Speech Analytics functionality will be available real-time to users of Twilio and SignalWire platforms.
1. Click here for instructions to access our live demo site.
2. If you are building a cool voice app and you are looking to test our APIs, click hereto sign up for a developer account and receive $50 in free credits
3. If you want to take Voicegain as your own AI Transcription Assistant to meetings, click here.
We are excited to announce a new Speech-to-Text (STT) API that works with AudioCodes VoiceAI Connect*. AudioCodes VoiceAI Connect (VAIC) enables enterprises to connect a bot framework and speech services, such as text-to-speech (TTS) and speech-to-text (STT), to the enterprises’ voice and telephony channels to power Voice Bots, conversational IVRs and Agent Assist use-cases.
With this new API, enterprises and NLU/Conversational AI platform companies can leverage the capabilities of AudioCodes VAIC with Voicegain as the ASR or STT engine for their contact center AI initiatives.
The two main use-cases in Contact Centers are (1) building Voice Bots (or voice-enabling a chatbot) and (2) building real-time Agent Assist.
While AudioCodes supports Cloud STT options from large players Microsoft, Google and Amazon, introducing Voicegain as an additional ASR option offers three key benefits to prospective customers. These benefits can be summarized as the 3 As - Accuracy, Affordability and Accessibility.
To get very high STT accuracy, companies now understand the importance of training the underlying acoustic models on application specific audio data. While it is necessary to have a reasonable out-of-the-box accuracy, building voice bots or extracting high quality analytics from voice data requires more than what is offered. Voicegain offers a full fledged training data pipeline and easy-to-use APIs to help speed up the building of custom acoustic models. We have demonstrated significant reduction in Word Error Rates even with a few hundred hours of client specific audio files.
Because AudioCodes VAIC makes it very easy two switch between various STT services, you can easily compare performance of Voicegain STT to any of the other STT providers supported on AudioCodes.
Voicegain offers disruptive pricing compared to the big 3 STT providers at essentially the same out-of-the-box accuracy. Our pricing is 40%-75% lower than the big 3 Cloud Speech-to-Text providers. This is especially important for real-time analytics (real-time agent assist) use case in contact centers as the audio/transcription volumes are very large. In addition to APIs, we also provide a white-label reference UI that contact centers can use to reduce the cost and time-to-market associated with deploying AI apps.
In addition to accessing STT as a cloud service, Voicegain can be deployed onto a Kubernetes cluster in a client's datacenter or in a dedicated VPC with any of the major cloud providers. This addresses applications where compliance, privacy and data control concerns prevent use of STT engines on public cloud infrastructure.
Connecting AudioCodes VAIC to Voicegain is done in 3 simple steps. They are:
1) Add Voicegain as the ASR/STT provider on VAIC. This is done through an API (provided by Audiocodes). In this step, you would need to enter a JWT token from Voicegain web console for authentication (instructions provided below).
2) Enter the web-socket entry URL for Voicegain ASR on VAIC. You can get this URL from Voicegain Web Console (instructions provided below)
3) Configure the Speech Recognition engine settings. This includes picking the right model and having the correct timeout and model sensitivity settings. This is done on the Voicegain Web Console (instructions to sign up provided below)
Please contact your Audiocodes customer success contact for Steps 1 & 2.
You would need to sign up for a developer account on Voicegain Web Console. Voicegain offers an open developer platform and there is no need to enter your credit card. We provide 300 minutes of free Speech-to-Text APIs access every month. You can test out our APIs and check out our accuracy.
After you sign up, please go to Settings-> API Security. The JWT Token required for Step 1 and the API entry URL for Step 2 are provided here.
Also you would need to pick the right acoustic model, set the complete timeout & sensitivity specified in Step 3. Please navigate Settings-> Speech Recognition -> ASR Transcription settings.
If you have any questions please email us at support@voicegain.ai
* VoiceAI connect is a product and trademark owned by AudioCodes.
The entire Voicegain Speech-to-Text/ASR platform and all the associated products - ranging from Web Speech-to-Text (STT) APIs, Speech Analytics APIs, Telephony Bot APIs and the MRCP ASR engine and our logging and monitoring framework - can deployed on "the Edge".
By "Edge" we mean that the core Deep Neural Network based AI models that convert speech/audio into text run exclusively on hardware deployed in a client datacenter. Or after this announcement they can also run on a compute instance in a Virtual Private Cloud. In either case, the Voicegain platform is "orchestrated" using the Voicegain Console which is a web application that is deployed on Voicegain cloud.
On the Edge, the Voicegain platform gets deployed as a container on a Kubernetes Cluster. Voicegain can also be accessed as a Cloud service if clients would not want to manage either server hardware or VPC compute instances.
Voicegain platform has always been deployable in a simple and automated way on hardware in a datacenter. Our clients procure compatible Intel Xeon based servers with Nvidia based GPUs. And they are able to install the entire Voicegain platform a few clicks from the Cloud Portal (see these videos for demonstration).
You can read about the advantages of this Datacenter type of Edge Deployment in our previous blog post. To summarize, these advantages are :
Now these benefits shall also be available for enterprise clients that use a Virtual Private Cloud on AWS to run a portion of their enterprise workloads.
Many enterprises have migrated several enterprise workloads to AWS Cloud infrastructure in order to benefit from the scale, flexibility and ease of maintenance. While moving these workloads to the Cloud, these enterprises largely prefer the private cloud offerings of AWS. e.g., using VPC network isolation, Site-to-Site VPN and dedicated compute instances. For these enterprises, ideally any new workload should be capable of being run inside their AWS VPC. In particular, if an enterprise already has dedicated AWS compute instances or hosts, they could realize all of the above 4 advantages of Edge Deployment by deploying into their dedicated AWS infrastructure.
Recently, anticipating interest of some of our customers we have performed extensive tests of complete deployment of our platform into AWS. Because Voicegain platform is Kubernetes based, there are essentially only two differences from deployment onto local on-premise hardware – these are:
Otherwise, the core of the deployment process is pretty much identical between on premise hardware and AWS VPC.
You can read the details involved in the AWS deployment process on Voicegain's github page.
If you are a developer building something that requires you to add or embed Speech-to-Text functionality (Transcription, Voice Bot or Speech Analytics in Contact Centers, analyzing meetings or sales calls, etc.), we invite you to give Voicegain a try. You can start by signing up for a developer account and use our Free tier. You can also email us at info@voicegain.ai.
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Read more →Donec sagittis sagittis ex, nec consequat sapien fermentum ut. Sed eget varius mauris. Etiam sed mi erat. Duis at porta metus, ac luctus neque.
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Read more →Interested in customizing the ASR or deploying Voicegain on your infrastructure?
Voicegain helps developers build awesome voice enabled apps by providing them with the most accurate, affordable and accessible Speech-to-Text platform.
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