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Average Ratings 0 Ratings

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ease
features
design
support

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Write a Review

Description

Bland Speech v3 is an innovative text-to-speech model that aims to generate voice audio that closely resembles that of a human, particularly in contexts like phone calls where overly polished speech may come off as inauthentic. This model captures essential human elements such as breathing, stutters, pauses, laughter, and throat-clearing by utilizing performance tags that are acted out rather than simply read. Users have the option to input their own scripts or utilize the Director feature to outline a conversation, allowing Bland to craft the dialogue, timing, and delivery prior to speech generation. Additionally, it offers voice cloning capabilities: a quick clone can be made from approximately 10 seconds of audio, whereas professional-grade cloning requires 30 minutes or more of verified audio, with users affirming that each voice belongs to them. Bland Speech can be accessed via a web studio and a single /v1/speak API endpoint, which employs bearer-key authentication for security. Audio is streamed through HTTP chunked transfer or WebSocket, returning PCM16 WAV files at a sample rate of 44.1 kHz, ensuring high-quality output for diverse applications. This versatility makes Bland Speech an essential tool for developers looking to enhance their audio experiences.

Description

TekSIP for Windows is a SIP Registrar as well as a SIP Proxy. TekSIP supports UDP (IPv4 & IPv6), TCP (IPv4 & IPv6), TLS, and WebSocket transports. Only commercial editions support Secure WebSocket and TLS. TekSIP has been tested on Microsoft Windows Vista and Windows 7/8/10/11 servers. TekSIP is a signaling service for WebRTC based SIP-based phones. TekSIP is compliant with RFC 3261 and RFC 3263. It also complies to RFC 3581, RFC 3891, RFC 3311 and RFC 3481. It supports NAT traversal, ENUM. Please refer to Readme for technical details on NAT traversal. You can choose the IP address that will be listened to and alternative SIP endpoints when making outgoing calls. You can also log the session details in a log and monitor registrations and active sessions in real-time. Windows Performance Monitor (Perfmon.exe) allows you to monitor the number of active sessions and registrations. TekSIP redirects calls to an alternate endpoint if the called endpoint is not available (Offline, busy ...). GUI can be used to terminate an active SIP session.

API Access

Has API Yes 

API Access

Has API Yes 

Screenshots View All

Screenshots View All

Integrations

Amazon Connect Yes 
Bland AI Yes 
Cal.com Yes 
Calendly Yes 
Five9 Yes 
Genesys Cloud CX Yes 
HubSpot CRM Yes 
Make Yes 
NiCE CXone Mpower Yes 
Notion Yes 
Pipedream Yes 
Salesforce Yes 
Slack Yes 
Talkdesk Yes 
TekRADIUS No 
Twilio Yes 
Zapier Yes 
iMessage Yes 

Integrations

Amazon Connect No 
Bland AI No 
Cal.com No 
Calendly No 
Five9 No 
Genesys Cloud CX No 
HubSpot CRM No 
Make No 
NiCE CXone Mpower No 
Notion No 
Pipedream No 
Salesforce No 
Slack No 
Talkdesk No 
TekRADIUS Yes 
Twilio No 
Zapier No 
iMessage No 

Pricing Details

$0.11 per minute
Free Trial Yes 
Free Version No 

Pricing Details

$239
Freeware version allows maximum 3 SIP endpoints registration and 3 simultaneous SIP sessions. Enterprise edition allows upto 100 simultaneous registration and SIP sessions. Following features are not supported in free edition;

Enterprise
* SDP Manipulation
* TLS Transport
* SMPP Gateway

SP
* Enterprise features
* External Authenticator
* SRTP -> RTP interworking
* WebRTC Media Gateway
* Unlimited number of simultaneous SIP registrations and sessions
* Route incoming calls to registered extensions
* IPv6 support
* RADIUS AAA
Free Trial Yes 
Free Version Yes 

Deployment

Web-Based Yes 
On-Premises Yes 
iPhone App No 
iPad App No 
Android App No 
Windows No 
Mac No 
Linux No 
Chromebook No 

Deployment

Web-Based No 
On-Premises Yes 
iPhone App No 
iPad App No 
Android App No 
Windows Yes 
Mac No 
Linux No 
Chromebook No 

Customer Support

Business Hours No 
Live Rep (24/7) No 
Online Support Yes 

Customer Support

Business Hours Yes 
Live Rep (24/7) Yes 
Online Support Yes 

Types of Training

Training Docs Yes 
Webinars No 
Live Training (Online) Yes 
In Person No 

Types of Training

Training Docs Yes 
Webinars No 
Live Training (Online) Yes 
In Person No 

Vendor Details

Company Name

Bland AI

Founded

2023

Country

United States

Website

www.bland.ai/speech

Vendor Details

Company Name

KaplanSoft

Founded

2011

Country

Turkey

Website

www.kaplansoft.com

Product Features

Product Features

Telephony

Auto-Dialer No 
Call Center Management No 
Call Monitoring Yes 
Contact Management No 
IVR / Voice Recognition No 
Inbound Reporting No 
Outbound Reporting No 
Predictive Dialer No 
Telemarketing Management No 
VoIP Yes 
Voice & Data Integration Yes 

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