Each service is subject to regional availability. Refer to the Developer Guide for details.

WebRTC Media

High-quality audio and video
Integrate the Amazon Chime SDK client libraries for JavaScript, iOS, Android or Windows into your applications to enable high-quality audio and video on WebRTC-enabled browsers and mobile operating systems. In a standard session, you can connect up to 250 participants with 48kHz stereo audio, 720p webcam video, and two 1080p screen share or content streams. In a high-definition session, you can connect up to 25 participants with 48kHz stereo audio, 1080p webcam video, and two 4K screen share o

Video simulcast and scalable video coding (SVC)
Create applications that automatically adapt to changing conditions with video simulcast and SVC. With simulcast enabled, clients upload multiple video streams from the same video source using different resolutions and bitrates. With SVC, clients upload a single video stream with multiple spatial and temporal layers. Clients displaying remote video tiles choose which stream or layer to download based on downlink bandwidth, number of video streams shared, and by prioritizing content share over webcam video streams.

Content sharing
Share pre-recorded audio and video, or a screen or application window with remote participants without the need to download or install an extension. All SDKs include embeddable screen viewing controls that provide the ability to zoom in and out or pan on the shared content. Callbacks track who is sharing and when the source of the sharing changes.

Active speaker
Track the active speakers through a periodic callback with a list of attendees sorted from most to least active. Use this information to enrich the application user experience with active talker labels, video tile highlighting or prioritization, and top talker information.

Volume notifications and mute controls
Use real-time notification callbacks to provide volume levels of attendees who are actively contributing audio for the session. Mute and unmute microphone actions and notifications are available to indicate the mute status for each attendee.

Device controller
Use the Device Controller module to manage the available audio input, audio output, and video input. Control permissions, preview, and selection of the devices used during the media session.

Optimized experience
Manage audio and video performance with both pre-built and customizable bandwidth policies controlling video bandwidth adaptation and video stream pausing. Builders can override these policies to get a more customized experience. The audio stream auto-reconnect feature provides more resilience to network interruptions.

Interactive Audience
Deliver real-time content over WebRTC to 10,000 audience members. Ideal for large meetings, webinars and lectures where audience members may want to interact with the presenters. The primary WebRTC media session can be replicated to up to 40 additional WebRTC media sessions for read-only consumption by audience members. Audience members can seamlessly transition to and from the primary session, without any WebRTC reconnection, to interact with presenters.

Attendee proximity
Pick the best AWS region to host each WebRTC media session based on where the attendees are joining from. Choose to host your WebRTC Media sessions in multiple control and data plane regions across the world, please check here for the latest available regions.

TURN support
WebRTC sessions include integrated Traverse Using Relay around NAT (TURN) service for corporate firewall and NAT transversal.

Workflows with Amazon EventBridge events
Leverage Amazon EventBridge events to trigger workflows based on session events like starting a call queue workflow when the session is created or when a specific attendee joins or leaves.

Video background replacement or blur
Help users increase their visual privacy with video background replacement or blur. Background replacement inserts an image behind the user, replacing the detected background. Alternatively, the background can be blurred with a choice of high-strength or low-strength blur.

Media recording

Record your WebRTC sessions to Amazon Simple Storage Service (Amazon S3), including audio, webcam video, content share, data messages, and transcription (if enabled).  Video streams can be recorded individually, or combined into single composited video stream. 

Audio analysis

Send your WebRTC session audio to Amazon Kinesis Video Streams to perform live analysis AWS ML services.  Send each attendees’ audio stream to a separate Kinesis Video Stream for individual analysis, or send a single mixed-audio stream with the meetings audio. 

Live connector
Send real-time video from your apps or websites to large audiences on your streaming platform. Amazon Chime SDK live connector is designed to transfer real-time video to streaming platforms like Facebook Live, YouTube Live, Amazon Elemental MediaLive, or Amazon Interactive Video Service (IVS), all with a few lines of code and without managing infrastructure. Using live connector, developers can enhance the streaming experience by pre-formatting video layouts and using machine learning to improve audio and video quality. Record video streams, then capture into a single file for video on demand (VoD) playback, offline consumption, or archiving in the cloud.

Speech enhancements

Noise reduction with Amazon Voice Focus
Reduce unwanted background noise with Amazon Voice Focus. Amazon Voice Focus is designed to reduce background noises such as dogs barking as well as foreground noises like keyboard typing to help listeners focus on the primary interaction. Amazon Voice Focus is included with WebRTC media sessions.

Echo reduction with double-talk improvement
Make every call clear. Our echo reduction technology uses machine learning to help keep acoustic echoes from recirculating while preserving the quality of the desired speech even in double-talk conditions.

Live transcription with Amazon Transcribe
Use live transcriptions generated by Amazon Transcribe or Amazon Transcribe Medical to overlay subtitles, build a transcript, or perform real-time content analysis. Transcription information can also be stored in your Amazon Simple Storage Service (Amazon S3) bucket with media capture for post-meeting processing.

Call analytics

Call analytics
Uncover call trends and enhance security and compliance in audio calls using machine-learning powered analytics. Call analytics reduces ML-based analytics deployment time from several months to a few days. Generate real-time voice analytics insights from speaker search and voice tone analysis, as well as transcriptions and insights such as call categorization and post-call summaries through pre-built integrations with Amazon Transcribe and Amazon Transcribe Call Analytics.
  • AWS AI integrations
    Use the ML workflow designer tool to ingest call audio from Amazon Chime Voice Connector to AWS AI services such as Amazon Transcribe and Amazon Transcribe Call Analytics with just a few clicks on the AWS console. Insights can be consumed both in real-time and after the completion of the call.
  • Speaker search
    Leverage real-time voice insights provided by machine-learning-powered speaker search. Speaker search uses a short voice sample from the call to perform a real-time search of your enrolled users and return the closest matches.
  • Voice tone analysis
    Estimate the sentiment expressed in how users say what they say. Voice tone analysis uses machine learning to estimate the sentiment expressed in a speech signal based on a combined analysis of linguistic and tonal information.
  • Recordings
    Record voice conversations to the S3 bucket of your choice for compliance or further analysis. Call recording includes an optional machine-learning-powered voice enhancement capability which reduces background noise and increases the fidelity of speech content in recordings.
  • Real-time alerts
    Insights become immediately actionable with real-time alerts. Configure alerts on one or a combination of insights, such as poor sentiment and specific phrases spoken.
  • Data lake integration
    Use your data lake for analysis across all your data. Store insights and call metadata in a data lake with data from other sources ready for SQL or other queries.

Public Switched Telephone Network (PSTN) Audio

Serverless Voice Applications
Build voice self-service applications, outbound voice notifications, and meeting join authentication using AWS Lambda serverless compute. Scale to thousands of simultaneous calls.  

Inbound Calling
Add Direct Inward Dial (DID/DDI) or toll-free telephone numbers to your voice applications. Route calls to an Amazon Chime SDK meeting, on-premises phone system, or any phone number.

Outbound Calling
Using an API, your application can make calls to landline or mobile numbers in over 100 countries with low, per-minute rates.
Call Recording
Monitor compliance, gather data for analysis, or simply save content for future use. Record calls to the Amazon Simple Storage Service (Amazon S3) bucket of your choice.

Phone Numbers for voice applications and meetings
Create or port your personalized dial-in numbers for the Amazon Chime SDK - based telephony applications and meetings. Toll-free and toll numbers are supported. You can also bring your own carrier and phone numbers and connect to your existing carrier using Amazon Chime Voice Connector (Session Initiation Protocol) SIP Trunking.
The native PSTN audio integration with  Amazon Polly lets you develop voice self-service menus, compose announcements, and play alphanumeric data with deep learning powered text-to-speech (TTS) synthesis.
Conversational Voice Experience
Build voice bots using Amazon Lex so that callers can use their voice to navigate phone system menus. You can build serverless phone self-service solutions and directly access Amazon Lex bots using a phone number provisioned within Amazon Chime SDK. The Amazon Chime SDK PSTN audio integration with Amazon Lex enables you to add natural language voice conversational experiences in your commercial or custom applications.  You can also use Amazon Chime Voice Connector and Amazon Chime SDK PSTN audio to access Amazon Lex bots in any enterprise communications or contact center application that uses Session Initiation Protocol (SIP).

SIP Trunking

SIP Trunking Setup
Amazon Chime Voice Connector supports standards-based Session Initiation Protocol (SIP); it is tested for compatibility with common SIP-based Private Branch Exchanges (PBXs), call centers, and Session Border Controllers (SBCs). With Amazon Chime Voice Connector, you can start using your SIP trunks in minutes using the Amazon Chime console or AWS SDK. There are no service orders needed and no trunk set-up charges.

T.38 Fax Support
Send and receive faxes from traditional fax machines connected to your phone system or via your SIP-enabled fax server.

Call Recording and Analytics
Capture real-time audio from SIP trunking and SIP-based media recording (SIPREC). Build real-time and post-call analytics using Amazon Transcribe, Amazon Comprehend, and other Machine Learning (ML) services.

Proxy Phone Sessions
Enable communications between two users via PSTN, without sharing personal phone numbers. Amazon Chime SDK proxy phone sessions support two-way voice or text communications sessions for up to 12 hours while protecting user privacy and keeping transactions within your service.

Phone Number Management and Porting
Manage your phone number inventory, order new numbers, and review pending transactions using the Amazon Chime console, AWS SDK, and AWS CLI. Keep your existing phone numbers by having us port them from your previous service provider. For more information please refer to Managing phone numbers in Amazon Chime SDK.

Call Health Monitoring
Use Amazon CloudWatch to get a unified view of voice calling operational health that includes call completion rates and voice quality metrics. You can use CloudWatch to detect anomalous behavior, set alarms, visualize logs and metrics, automate actions, and troubleshoot issues to keep your voice calling running smoothly.

Security and Authentication
You can use encryption to secure calls between your phone system and AWS. Amazon Chime Voice Connector supports TLS to encrypt SIP signaling and SRTP to encrypt media streams. You can optionally enable SIP digest authentication to authenticate outbound calls from your phone system.

Quality and Reliability
Amazon Chime Voice Connector supports fault-tolerant, cross-region inbound and outbound call routing to and from your phone system. In the event of loss of connectivity between an AWS region and your phone system, or if there is an issue with the Amazon Chime Voice Connector service in a region, calls will route to the other Amazon Chime Voice Connectors.

Cost-effective Network Access
You can access your Amazon Chime Voice Connector over AWS Direct Connect or your Internet connection. Using AWS Direct Connect, you can establish private connectivity between AWS and your data center, office, or colocation environment, which can help provide a more consistent network experience than Internet-based connections. With the elastic scaling, you can meet all your call volume needs, including unexpected or seasonal spikes, and you no longer need to purchase bursting capacity or over-provision concurrent call paths.


Amazon Lex Integration
The Amazon Chime SDK integrates natively with Amazon Lex, allowing builders to more simply create chatbots in web and mobile applications. Through the Amazon Chime SDK, builders can also add automated redaction of PII, sentiment analysis, persistence or escalation of a bot conversation to a human.

Up to 100k Members per Channel
Organize users in channel sizes ranging from one-on-one conversations to broadcast channels with up to 100k members.

Power workflows with machine learning and cross-application messaging
Channel flows enable you to process messages in channel processors (AWS Lambda functions) before they are delivered. Channel flows can be used to remove sensitive data such as government ID numbers, phone numbers, or profanity from messages before they are delivered. Channel flows can also be used to perform functions like aggregation of responses to a poll before sending results back to participants or sending messages to recipients via SMS or other popular messaging applications.

Public and Private Channels
Create public channels that any user can find and join, or private channels that require members or moderators to add new members.

Message history
Message history is available to your users for as long as you choose.

Message redaction
Allow moderators and users to redact messages containing sensitive data so they are no longer accessible by users.

E dit messages
Enable users to edit any message they send.

Moderation tools
Using the Amazon Chime SDK, user channel moderators can redact inappropriate messages, kick users out of channels, and ban users to prevent them from rejoining.

Include attachments stored in Amazon S3 or your choice of storage location as message metadata.

Message retention policies and deletion
Implement retention policies, if required. Automatically delete any messages after a retention period you choose, ranging from 1 day to 15 years. APIs are also available to delete individual messages and channels, permanently removing them from Amazon Chime.

Unlimited Connections
There are no hard limits on the number of users that can use Amazon Chime SDK messaging.

Push Notifications
Send push notifications through Amazon Pinpoint to users for new messages or calls even when they do not have your app open.

Use existing Identity Provider
Choose to use Amazon Cognito for user management and authentication, or use your existing identity provider.

Control messages
Use control messages, up to 30 bytes, for features like typing indicators, message effects, or presence.

Streaming export of chat data through Amazon Kinesis
Export Amazon Chime SDK chat data through Amazon Kinesis to provide search, archive data, process data, or train ML models.

React UI Components

Use React UI components for common messaging elements.

Search for conversations by members or unique ID such as an appointment ID or game ID.

Automatically pre-fetch information to display a rich chat UI to users when clients load including surfacing channels that require the users attention first.

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