Average Ratings 0 Ratings

Total
ease
features
design
support

No User Reviews. Be the first to provide a review:

Write a Review

Average Ratings 0 Ratings

Total
ease
features
design
support

No User Reviews. Be the first to provide a review:

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

API Access

Has API

Screenshots View All

Screenshots View All

Integrations

Amazon Connect
Bland AI
Cal.com
Calendly
Five9
Genesys Cloud CX
HubSpot CRM
Make
NiCE CXone Mpower
Notion
Pipedream
Salesforce
Slack
Talkdesk
TekRADIUS
Twilio
Zapier
iMessage

Integrations

Amazon Connect
Bland AI
Cal.com
Calendly
Five9
Genesys Cloud CX
HubSpot CRM
Make
NiCE CXone Mpower
Notion
Pipedream
Salesforce
Slack
Talkdesk
TekRADIUS
Twilio
Zapier
iMessage

Pricing Details

$0.11 per minute
Free Trial
Free Version

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
Free Version

Deployment

Web-Based
On-Premises
iPhone App
iPad App
Android App
Windows
Mac
Linux
Chromebook

Deployment

Web-Based
On-Premises
iPhone App
iPad App
Android App
Windows
Mac
Linux
Chromebook

Customer Support

Business Hours
Live Rep (24/7)
Online Support

Customer Support

Business Hours
Live Rep (24/7)
Online Support

Types of Training

Training Docs
Webinars
Live Training (Online)
In Person

Types of Training

Training Docs
Webinars
Live Training (Online)
In Person

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
Call Center Management
Call Monitoring
Contact Management
IVR / Voice Recognition
Inbound Reporting
Outbound Reporting
Predictive Dialer
Telemarketing Management
VoIP
Voice & Data Integration

Alternatives

Fish Audio Reviews

Fish Audio

Hanabi AI

Alternatives

Inworld TTS Reviews

Inworld TTS

Inworld
TekIVR Reviews

TekIVR

KaplanSoft
Voxtral TTS Reviews

Voxtral TTS

Mistral AI