WebRTC

Infobip WebRTC offers a secure way of establishing real-time communication over the web and within mobile or web applications. This helps you focus on user experience and business logic within your app.

It is powered by Web Real-Time Communication (WebRTC) technology, the leading real-time communication standard built into more than a billion devices.

You can use WebRTC as a standalone product or combined with different voice capabilities. To enable WebRTC, contact your dedicated Account Manager or our Sales team.

Infobip currently supports several types of one-to-one calls: 

• Web browser to web browser audio and video calls.
• Mobile application to mobile application audio and video calls.
• Web browser to mobile application audio and video calls and vice versa.
• Web browser and mobile application audio calls to phone calls and vice versa.
• Web browser and mobile application voice calls to SIP server.

WebRTC calls support Dynamic Destination Resolving.

Additionally, WebRTC can be recorded. Recording is available as an add-on so please contact your dedicated Account Manager to set it up. Learn more about Recording.
 

Here's how WebRTC works.

Calls between end users in your app need to be private. This is where Infobip WebRTC comes into place. Calls are made over VoIP without using any phone networks. However, if a phone network service is needed, end users can still be connected thanks to the interoperability of the Infobip Voice platform.

You can integrate WebRTC within your app using the Infobip RTC SDKs.

Features

Technical Requirements 

Here are the technical details you need to know before you start using WebRTC.

Bandwidth Requirements

The approximate data rate for each stream of WebRTC:

Codecs

Platform Compatibility

For supported Android versions, see the supported list.

For supported iOS versions, see iOS system requirements.

For supported browsers and OS, see browser compatibility list.

Feature Platform Compatibility

The table below depicts the webRTC SDK features available on different supported platforms.

Getting Started

To start using WebRTC, you need to have an Infobip account and WebRTC enabled as a service on that account. Log in or create an account to continue.

To enable WebRTC, contact your dedicated Account Manager or contact our Sales team.

To get started and set up WebRTC, follow the instructions below.

Step 1 - Create your application and define the type. You can create the application in the web interface or via API.

Step 2 (optional) - Set up push notifications.

Step 3 - Set up SDKs for your application project.

Depending on your use case, these could be optional steps:

Step 4 - Configure SIP trunk in case you want to forward the calls to your SIP platform.

Step 5 - Configure Dynamic Destination Resolving.

To enable WebRTC, go through these 5 steps.

Configure Application

The WebRTC application can be configured via API or in the web interface. Follow the instructions below to create and configure your application over web interface.

1. Navigate to the Channels and Numbers module located at the bottom left > WebRTC > Applications > ADD NEW.

2. Name your application and add the description.
   

Set Up Push Notifications

This is optional. However, we highly recommend you set this up if you want to place calls to end users within the application.

For iOS:

• Upload APNs Certificate.

• If the certificate is encrypted, enter the password.

For Android:

• Enter the FCM Server API Key.

Set Up Application Options

Here you can define in which modes the application will be used. Setup options correspond to SDK method options.

App to App

• Select this option if you plan to use your application for establishing WebRTC calls. End users will be connected via the Internet within your app.

• SDK Call method (Web, Android, iOS) is used.

App to Conversations

• Select this option if you plan to use your application for establishing WebRTC calls to Infobip Conversations.

• SDK CallConversations method (Web, Android, iOS) is used.

App to SIP

This step is optional. Required only if you will use Forwarding to your SIP platform.

• Select this option if you plan to use your application for forwarding WebRTC calls to your SIP platform. Your platform should support SIP protocol and should be reachable over the public Internet via static and public IP address.

• Link the application to the correct SIP Trunking configuration. 

• SDK CallSIP method (Web, Android, iOS) is used

• Currently, Video calls are not supported with this option.

App to Dynamic Destination Resolving

• Select this option if not all WebRTC call parameters are known when the call is initiated. DDR is required when you want to route incoming phone calls to the application user.

• Link the application to the correct Dynamic Destination Resolving configuration.

• SDK CallDynamicDestinationRequest method (Web, Android, iOS) is used.

• Learn more about Dynamic Destination Resolving.

App to Phone

• Select this option if you plan to use your application for connecting WebRTC calls to phone numbers. 

• Lease a Voice number if you want to have a dedicated caller ID. Otherwise, the caller ID for this call will be a random number designated for that connection.

• SDK CallPhoneNumber method (Web, Android, iOS) is used.

FORWARD WEBRTC CALLS TO SIP PLATFORM

In case you would like to receive calls from the web browser and mobile applications directly to your SIP platform, configure SIP trunking first.

Create SIP Trunking Configuration

1. Navigate to the Channels and Numbers module located at the bottom left > WebRTC > SIP Trunking > ADD NEW.

2. Select SIP Trunking from the module tabs.

3. Enter the following parameters and click CREATE SIP TRUNK:

• IP Address - Enter the IP address of the SIP Server to which calls should be forwarded to
• Port - Define the TCP/UDP port used for SIP communication
• Data Center - Select the Infobip data center closest to your platform location

 The same SIP trunking configuration can be used for placing the calls over our Voice platform. Learn more about Voice SIP trunking.

After you're done with the SIP trunking configuration, you can go back to the WebRTC application and enable App to SIP. This is to forward the calls to your SIP platform.

Dynamic Destination Resolving

People's privacy is crucial. Brands need to ensure that they are keeping all personal data safe. This includes but is not limited to, names and phone numbers.

Infobip Dynamic Destination Resolving feature in WebRTC calls enables your end users to talk to each other in full privacy and protect their identities. 

At the same time, you avoid off-platform transactions and have full control of the communication flowing in to allow for advanced routing and cost optimization.

This is what a high-level overview looks like:

Your end user can initiate a call from the application or call the Voice number. When the call reaches the Infobip platform, the platform calls HTTP API endpoint that you have specified when configuring your application. On your platform via API, you decide what you want to do with the call.

To set up Dynamic Destination Resolving, specify the HTTP API URL that the Infobip platform needs to call.

Here's how to set up Dynamic Destination Resolving configuration.

Create Dynamic Destination Resolving Configuration

1. Navigate to the Channels and Numbers module located at the bottom left > WebRTC > Dynamic Destinations > ADD NEW.

2. Enter the following parameters and click CREATE DYNAMIC DESTINATION.

• Authorization header (optional) – Enter the Authorization request header as the general HTTP authentication framework. That is, if you want to have it included in the request we send toward your platform. It adds an additional layer of security to intrap-latform communication.
• Callback URL (API endpoint) – Define where you would like us to send the request to.
• Status URL (API endpoint) – Define where would you like to receive status reports once the call is finished.

3. You can go back to the WebRTC application and enable App to Dynamic Destination Resolving to use this configuration.

4. Align your platform with the Infobip platform when receiving incoming calls. To set this up, refer to Incoming Calls with Dynamic Destination Resolving. 

Incoming Calls with Dynamic Destination Resolving

Incoming calls can be initiated by:

• WebRTC calls from end users using a mobile or web application,

• Placing a phone call to Voice number.

In both cases, when the call reaches the platform, we will send a POST request to your API endpoint.

Depending on the type of the incoming call, the request will contain different information:

If the call originated from the application, it will contain source and customData fields. 

• Source is the identity (unique identifier) of the user who initiated the call.

• CustomData can be passed through Web, Android and iOS SDKs when calling our callDynamicDestination method.
  CustomData can be used to collect information for routing.

If the incoming call was a phone call it will contain the source and destination fields.

• Source is the phone number of the person who made the call.

• Destination is the DID number that the phone call has been placed to.

Request example:
(call originating from WebRTC calls)

{ 
"source": "Alice", 
"customData": { 
	"myUserId": "myBob" 
} 
}

Request example:
(call originating from an actual phone number)

{ 
"source": "41793026727", 
"destination": "41793026731" 
}

In the response, define where the call will be delivered. Specify the action and destination fields.

• With the action, you define whether the call will terminate as WebRTC call towards the end user in the application or if it will be delivered to a phone number. Set the action field value to call or call_phone_number.  

• With the destination field, you provide the information which end user or phone number to forward the call to. Optional field in the response is source but it is necessary when you want to guarantee anonymity to your customers. If you don't' set it up, the original source that was received with the call will be used.

Response example:
(call terminating to WebRTC calls)

{
"action": "call",
"source": "AnotherAlice",
"destination": "AnotherBob”
}

Response example:
(call terminating to an actual phone number)

{ 
"action": "call_phone_number", 
"source": "41793026728", 
"destination": "41793026732" 
}

If you configured the status URL after the call has been completed, the platform will send another HTTP POST request to that endpoint. It will contain general information regarding the call status.

Request example: 

{ 	
"callId": "000a15f9-187b-425e-8f16-63acb54eeb4c", 
// obtained by calling "id()" method on "Call" object in our SDKs 
"source": "Alice", 
"destination": "Bob", 
"status": "finished", // or "failed" 
"duration": 15 
}

The Infobip platform doesn't expect a response. Your platform needs to return the 200 OK status so we know you have successfully received the information.

Video Calls

Video calls are an integral part and an embedded functionality with WebRTC calls. While your end users are on the call, they can either turn the video on or off.

Media Switch

Seamless switch between audio and video calls. You can start the call as video or audio and then switch video on and off as needed during your call.

Screen Share

Share screen during calls.

Types of sharing screen:

• Full - Share the entire screen.

• In-app - Select one app you want to share during your call.

Full Screen Share is available on browsers (see Compatibility list) and Android OS, while In-app Screen Share is available on iOS.

Network Events

Inform your end users about network quality or networking issues during their WebRTC calls. Offer a better user experience using our network quality events feature.

If network problems occur during a call between two users, the platform will inform the user experiencing this issue that their network is too slow to support a high-quality call.

At the same time, the other user will be informed about their peer's network quality issues, so they are aware the problem lies on the other side of the call.

Depending on the platform where you'll incorporate this feature, use the following links:

• JavaScript SDK for browsers

• iOS SDK

• Android SDK

Conferencing

The Conferencing feature enables you to create a conference call in WebRTC.

It is available via SDKs:

• Audio conferencing - Available in browsers and mobile applications. 

• Video conferencing - Available in browsers and mobile applications.

Depending on the platform where you'll incorporate the Conferencing feature, use the following links:

• JavaScript SDK for browsers
• iOS SDK
• Android SDK

Check out the table below for more information about the WebRTC feature availability: 

Smart video distribution refers to the distribution of the participants' video stream in a conference. Streams are ordered based on the user activity and display 3 web or 1 mobile users' cameras in high quality (720p).

Recording WebRTC Calls

You can record video and audio calls with WebRTC calls. 
You can record all video calls. Recording is available as an add-on so please contact your dedicated Account Manager.

To turn on the Recording feature, set the capability option for recording to ALWAYS or ON_DEMAND when creating the token for the session. After the recorded audio or video call has ended, recorded files will be available to you. You may search, download and/or delete them over API or via web interface.

Additionally, we provide the option of turning the recording on and off for each individual call, overriding the default setup option. When configuring this option per call, you can specify the type of media you want to record - audio, video, or both.

For more details on managing files over the web interface as well as detailed information on recording in general, refer to Recording.