Meta Developers

As of April 2023 the Rooms API is deprecated and is no longer included in the list of available Platform APIs for integration. For more information click here.

Third-party solutions, such as Photon, can be used as an alternative to implement Rooms in your app. For an example of how Photon, and other third-party solutions integrate with the Platform API, check here.

Use the Platform VoIP service to add voice chat to your app.

The VoIP service transmits PCM (pulse-code modulation) data between users, which the Platform SDK decodes to audio.

VoIP is peer-to-peer so your app will be limited by bandwidth in most cases. In general, the VoIP service becomes unstable if you have 12 or more connections. You should have 8 or less 8 connections on Rift and 4 or less on mobile apps. You may be able to support more if you use push to talk or some proximity-type muting.

Note

VoIP data may be routed through relay servers located inside Meta's infrastructure when connecting to client devices that are behind network address translation (NAT), if a connection cannot be established from their detected external address. The VoIP data passing through a relay server is not inspected any more than is necessary to forward to the recipient; nor is it retained.

This is a Platform SDK feature requiring Data Use Checkup

To use this or any other Platform SDK feature, you must complete a Data Use Checkup (DUC). The DUC ensures that you comply with Developer Policies. It requires an administrator from your organization to certify that your use of user data aligns with platform guidelines. Until the app review team reviews and approves your DUC, platform features are only available for test users.

Integrating VoIP

To integrate Application VoIP in your Native application, follow the steps below. The methods can be called from your client app.

First, if you are releasing your application for the Quest, you need to ensure the RECORD_AUDIO permission is added to your APK package’s manifest.

Send a VoIP connection request

Use the following method to initiate a VoIP connection request with a specified user.

ovr_Voip_Start()

Accept a VoIP connection request

Use the following method to accept a VoIP connection request.

ovr_Voip_Accept()

Retrieve the maximum size of the buffer

To retrieve the size of the internal ringbuffer used by the VoIP system in elements, use the following method. This size is the maximum number of elements that can ever be returned by ovr_Voip_GetPCM or ovr_Voip_GetPCMFloat.

ovr_Voip_GetOutputBufferMaxSize()

Retrieve PCM data from a user (16 bit fixed point)

The following method retrieves all available samples of voice data from a specified user and copy them into the outputBuffer. The VoIP system will generate data at roughly the rate of 480 samples per 10ms. This function should be called every frame with 50ms (2400 elements) of buffer size to account for frame rate variations. The data format is 16 bit fixed point 48khz mono.

ovr_Voip_GetPCM()

Retrieve PCM data from a user (32 bit floating point)

Retrieve all available samples of voice data from a specified user and copy them into the outputBuffer. The VoIP system will generate data at roughly the rate of 480 samples per 10ms. This function should be called every frame with 50ms (2400 elements) of buffer size to account for frame rate variations. The data format is 32 bit floating point 48khz mono.

We do not recommend using floating point unless necessary. This operation requires additional resources when compared to the integer based process.

ovr_Voip_GetPCMFloat()

Retrieve the number of PCM data files available

Use the following method to retrieve the current number of audio samples available to read from a specified user. This function is inherently racy, it’s possible that data can be added between a call to this function and a subsequent call to ovr_Voip_GetPCM or ovr_Voip_GetPCMFloat.

ovr_Voip_GetPCMSize()

Retrieve PCM data with a timestamp from a user (16 bit fixed point)

Like ovr_Voip_GetPCM, this function copies available audio samples from a specified user into a buffer. Along with the audio samples, this function also stores the timestamp of the first sample in the output parameter timestamp. This timestamp can be used for synchronization, see ovr_Voip_GetSyncTimestamp for more details. The data format is 16 bit fixed point 48khz mono.

This function may return data early, even if there’s more data available, to keep the batch of audio samples returned with a single timestamp small. For example, if 30ms worth of audio is in the buffer, this function may return 480 samples (10ms) each time it’s called. Therefore, it’s recommended to call this as long as there’s data in the buffer (i.e. the function returns a non-zero result).

ovr_Voip_GetPCMWithTimestamp()

Retrieve PCM Data with a timestamp from a user (32-bit floating point)

Like ovr_Voip_GetPCMFloat, this function copies available audio samples from a specified user into a buffer. Along with the audio samples, this function also stores the timestamp of the first sample in the output parameter timestamp. This timestamp can be used for synchronization, see ovr_Voip_GetSyncTimestamp for more details. The data format is 32 bit floating point 48khz mono.

This function may return data early, even if there’s more data available, in order to keep the batch of audio samples with a single timestamp small. For example, if there’s 30ms worth of audio in the buffer, this function may return 480 samples (10ms) each time it’s called. Therefore, it’s recommended to call this as long as there’s data in the buffer (i.e. the function returns a non-zero result).

We do not recommend using floating point unless necessary. This operation requires additional resources when compared to the integer based process.

ovr_Voip_GetPCMWithTimestampFloat()

Retrieve the sync timestamp for the sender

Timestamps associated with audio frames are implicitly transmitted to remote peers; on the receiving side, they can be obtained by using ovr_Voip_GetPCMWithTimestamp. ovr_Voip_GetSyncTimestamp is used to fetch those timestamps on the sending side. An application can insert the value returned by this function into each data packet and compare it to the value returned by GetPCMWithTimestamp on the receiving side to determine the ordering of two events (sampling audio and composing a data packet).

Note: The timestamp is generated by an unspecified clock and won’t represent wall-clock time. Use ovr_Voip_GetSyncTimestampDifference to determine the difference between two timestamps in microseconds. This function assumes that a voice connection to the user already exists; it returns 0 if that isn’t the case.

ovr_Voip_GetSyncTimestamp()

Returns a timestamp used for synchronizing audio samples sent to the user with an external data stream.

Retrieve the difference (in Âµs) between two timestamps

Use this method to retrieve the calculated difference between two sync timestamps, in microseconds. returned by ovr_Voip_GetSyncTimestamp, ovr_Voip_GetPCMWithTimestamp, or ovr_Voip_GetPCMWithTimestampFloat.

ovr_Voip_GetSyncTimestampDifference()

The return value will be negative if lhs is smaller than rhs, zero if both timestamps are the same, and positive otherwise. The absolute value of the result is the time in microseconds between the two sync timestamps.

Retrieve the mute state of the microphone

This method checks if the microphone has been muted.

ovr_Voip_GetSystemVoIPMicrophoneMuted()

Retrieve the VoIP system status

The following method retrieves the status of System VoIP. System VoIP is the service used by Meta Quest Parties. See the Mic Switcher page for information about managing the microphone.

ovr_Voip_GetSystemVoipStatus()

Set the microphone filter callback

Set a callback that will be called every time audio data is captured by the microphone.

ovr_Voip_SetMicrophoneFilterCallback()

The callback function must match this format:

void filterCallback(int16_t pcmData[], size_t pcmDataLength, int frequency, int numChannels);

The pcmData param is used for both input and output. pcmDataLength is the size of pcmData in elements. numChannels will be 1 or 2. If numChannels is 2, then the channel data will be interleaved in pcmData. frequency is the input data sample rate in hertz.

Mute/unmute the microphone

This function is used to enable or disable the local microphone. When muted, the microphone will not transmit any audio. VoIP connections are unaffected by this state. New connections can be established or closed whether the microphone is muted or not. This can be used to implement push-to-talk, or a local mute button. The default state is unmuted.

ovr_Voip_SetMicrophoneMuted()

Set the VoIP output sample rate

Set the output sample rate. Audio data will be resampled as it is placed into the internal ring buffer.

ovr_Voip_SetOutputSampleRate()

End the VoIP connection

Use the following method it end a VoIP session with a specified user. Note that a muting functionality should be used to temporarily stop sending audio; restarting a VoIP session after tearing it down may be an expensive operation.

ovr_Voip_Stop()

Setting VoIP Options – DTX and Bit Rate

You can optionally configure your app to use discontinuous transmission (DTX) and set the bit rate for VoIP connections.

DTX allows for temporary microphone muting when there’s no voice input. This improves efficiency because a person is usually talking less than half the time in a typical two-way conversation. This can conserve battery life and free up the channel. By default, DTX is disabled.

Setting the bit rate controls the network usage when transmitting on VoIP connections. The higher the bit rate, the better the audio quality, but more network resources are used. A lower bit rate will conserve network resources. By default, the bit rate is 32,000 bits per second.

Once these options have been configured, they apply only to new connections. Existing connections must be closed and reconnected for the new settings to be applied.

To configure these options in a Native application, follow these steps:

Create an options handle with ovr_VoipOptions_Create.
Configure the desired options using the handle with ovr_VoipOptions_SetCreateNewConnectionUseDtx and ovr_VoipOptions_SetBitrateForNewConnections.
Enable the newly configured options with ovr_Voip_SetNewConnectionOptions.
Destroy the handle with ovr_VoipOptions_Destroy.

These functions are described below.

An example flow of calls in a Native application would look like this:

ovrVoipOptionsHandle options = ovr_VoipOptions_Create();
ovr_VoipOptions_SetCreateNewConnectionUseDtx(options, ovrVoipDtxState_Enabled);
ovr_VoipOptions_SetBitrateForNewConnections(options, ovrVoipBitrate_B64000);
ovr_Voip_SetNewConnectionOptions(options);
ovr_VoipOptions_Destroy(options);

Create a handle to set VoIP options:

Creates a handle used to set options.

ovr_VoipOptions_Create()

Set DTX usage

The following method is used to enable or disable DTX. Passing the options handle and a value of ovrVoipDtxState_Enabled enables DTX on all new VoIP connections. It does not apply to existing connections, which must be closed and then reconnected for DTX to be enabled. Passing the options handle and a value of ovrVoipDtxState_Disabled similarly disables DTX on all new connections.

ovr_VoipOptions_SetCreateNewConnectionUseDtx()

In Unity, this is set to ovrVoipDtxState.Enabled or ovrVoipDtxState.Disabled.

By default, apps will have DTX disabled.

Set the bit rate for new connections

Used to set the bit rate on all new VoIP connections. The higher the bit rate, the better the audio quality, at the expense of network usage. It does not apply to existing connections, which must be closed and then reconnected for the new bit rate to take effect.

ovr_VoipOptions_SetBitrateForNewConnections()

To change the bit rate, pass the options handle and one of the following values:

+ ovrVoipBitrate_B16000
+ ovrVoipBitrate_B24000
+ ovrVoipBitrate_B32000  (default)
+ ovrVoipBitrate_B64000
+ ovrVoipBitrate_B96000
+ ovrVoipBitrate_B128000

Enable the newly configured options

Enables the options set on all new connections.

ovr_Voip_SetNewConnectionOptions()

Destroy the VoIP options handle**

Destroys the options handle.

ovr_VoipOptions_Destroy()

In addition, there are several functions related to these options.

Check if a connection is using DTX

Checks if a connection is using DTX. If a connection exists and both sides have DTX enabled, this returns ovrVoipDtxState_Enabled (Unity: ovrVoipDtxState.Enabled). If a connection exists and only one or neither side has it enabled, this returns ovrVoipDtxState_Disabled (Unity: ovrVoipDtxState.Disabled). If there’s no existing connection, this returns ovrVoipDtxState_Unknown (Unity: ovrVoipDtxState.Unknown).

ovr_Voip_GetIsConnectionUsingDtx()

Check the bit rate for users

These functions return an ovrVoipBitrate value specifying the VoIP bit rate of the remote and local users respectively. The bit rates do not have to be the same. For example, if the remote bit rate is very high and the local bit rate is very low, the local user will hear high quality audio but send low quality audio. If there is no existing connection, this returns ovrVoipBitrate_Unknown

For a local remote user:

ovr_Voip_GetRemoteBitrate()

For a remote user:

ovr_Voip_GetLocalBitrate()

Implementation Overview

This section will walk you through the basic process of implementing VoIP in your Native app. Before integrating VoIP, please review the Parties page for information about potential conflicts with users who launch your app while participating in a party. Neglecting to do so will result in a poor user experience.

The general process of implementing VoIP is:

Send the connection request. The user initiating the connection request calls ovr_Voip_Start with the Id of the user they would like to connect with. Information about retrieving user id’s can be found or the Users, Friends, and Relationships.
Accept the connection request. The user on the receiving end of the will be listening for notifications on their message queue. After handling the request, the receiving user accepts by calling ovr_Net_Accept with the peerID of the request.
Listen for the state change. Both users will be listening for the notification that the connection has been established by listening for ovrMessage_Notification_Voip_StateChange. Once this is received the users can begin exchanging data.
Transmit PCM data. The connected microphone will begin capturing the voice data and the SDK will convert to PCM data and transmit to the user the connection was established with.
Determine the size of the buffer. Call ovr_Voip_GetOutputBufferMaxSize to determine the max output size, or the maximum number of elements that can be retrieved.
Retrieve PCM data for processing and playback. There are a number of methods you can use to retrieve the PCM data depending on your application’s use-case. Before you retrieve any data, call ovr_Voip_GetPCMSize to determine the size of the data in the buffer. Then call one of the following methods to retrieve the data. One of these methods needs to be called for each connection, PCM data will only be returned for a single user connection.
1. ovr_Voip_GetPCM - Retrieves the PCM data in 16 bit fixed point 48khz mono format. ovr_Voip_GetPCMWithTimestamp retrieves the same data with an additional timestamp that can be used for synchronization.
2. ovr_Voip_GetPCMFloat - Retrieves the PCM data in 32 bit floating point 48khz mono. ovr_Voip_GetPCMWithTimestampFloat retrieves the same data with an additional timestamp that can be used for synchronization. Using floating point is not recommended unless necessary.
(Optional) After retrieving the PCM data, you may wish to spatialize the audio playback. Information about spatializing audio can be found in the Audio SDK documentation.
(Optional) At any time either user may mute their microphone by calling ovr_Voip_SetMicrophoneMuted. Calling the same function will unmute the microphone. Call ovr_Voip_GetSystem VoIPMicrophoneMuted to check the mute state of the microphone.
End the connection. Terminate the connection between users at any time, call ovr_Voip_Stop to close.

Notifications and the Message Queue

As mentioned above, the VoIP service uses notifications to update your app on the status of the connection and service. Review the Requests and Messages page for more information about the message queue and notifications.

For example, a native application may listen for state changes using:

case ovrMessage_Notification_Voip_StateChange: {
    ovrNetworkingPeerHandle netPeer = ovr_Message_GetNetworkingPeer(message);
    ovrPeerConnectionState netState = ovr_NetworkingPeer_GetState(netPeer);
    printf(
        "User: %llu, voip state %s\n",
        ovr_NetworkingPeer_GetID(netPeer),
        ovrPeerConnectionState_ToString(netState)
    );
}
Break;

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