Class: BaseRtpEndpoint

(abstract) core/abstracts.BaseRtpEndpoint()

Base class for the implementation of RTP-based communication endpoints.

All endpoints that rely on the RTP protocol, like the :rom:cls:`RtpEndpoint` or the :rom:cls:`WebRtcEndpoint`, inherit from this class. This endpoint provides information about the Connection state and the Media state, which can be consulted at any time through the core/abstracts.BaseRtpEndpoint#getMediaState and the core/abstracts.BaseRtpEndpoint#getConnectionState methods. It is also possible subscribe to events fired when these properties change:

  • ConnectionStateChangedEvent: This event is raised when the connection between two peers changes. It can have two values:
    • CONNECTED
    • DISCONNECTED
  • MediaStateChangedEvent: This event provides information about the state of the underlying RTP session. Possible values are:
    • CONNECTED: There is an RTCP packet flow between peers.
    • DISCONNECTED: Either no RTCP packets have been received yet, or the remote peer has ended the RTP session with a BYE message, or at least 5 seconds have elapsed since the last RTCP packet was received.

    The standard definition of RTP (RFC 3550) describes a session as active whenever there is a maintained flow of RTCP control packets, regardless of whether there is actual media flowing through RTP data packets or not. The reasoning behind this is that, at given moment, a participant of an RTP session might temporarily stop sending RTP data packets, but this wouldn't necessarily mean that the RTP session as a whole is finished; it maybe just means that the participant has some temporary issues but it will soon resume sending data. For this reason, that an RTP session has really finished is something that is considered only by the prolonged absence of RTCP control packets between participants.

    Since RTCP packets do not flow at a constant rate (for instance, minimizing a browser window with a WebRTC's RTCPeerConnection object might affect the sending interval), it is not possible to immediately detect their absence and assume that RTP session has finished. Instead, there is a guard period of approximately 5 seconds of missing RTCP packets before considering that the underlying RTP session is effectively finished, thus triggering a MediaStateChangedEvent = DISCONNECTED event.

    In other words, there is always a period during which there might be no media flowing, but this event hasn't been fired yet. Nevertheless, this is the most reliable and useful way of knowing what is the long-term, steady state of RTP media exchange.

    The ConnectionStateChangedEvent comes in contrast with more instantaneous events such as MediaElement's MediaFlowInStateChangedEvent and MediaFlowOutStateChangedEvent, which are triggered almost immediately after the RTP data packets stop flowing between RTP session participants. This makes the MediaFlow events a good way to know if participants are suffering from short-term intermittent connectivity issues, but they are not enough to know if the connectivity issues are just spurious network hiccups or are part of a more long-term disconnection problem.

Bitrate management

Bandwidth control for the video component of the media session is done here. Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

  • Recv bandwidth: Used to request a remote sender that its media bitrate is within the requested range.
    • MinVideoRecvBandwidth: Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.
      • Unit: kbps (kilobits per second).
      • Default: 0.
      • Note: The actual minimum value is 30 kbps, even if a lower value set.
    • MaxAudioRecvBandwidth and MaxVideoRecvBandwidth: Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
      • Unit: kbps (kilobits per second).
      • Default: 0 (unlimited).
  • Send bandwidth: Used to control bitrate of the outbound media stream sent to remote peers. It is important to keep in mind that outbound bitrate ultimately depends on network and remote peer capabilities.
    • MinVideoSendBandwidth: Minimum outbound bitrate sent by this endpoint.
      • Unit: kbps (kilobits per second).
      • Default: 100.
      • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.
    • MaxVideoSendBandwidth: Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.
      • Unit: kbps (kilobits per second).
      • Default: 500.
      • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
        Note that this might have a bad effect if more than one stream is running (as all of them would try to raise the video bitrate indefinitely, until the network gets saturated).
    • RembParams.rembOnConnect: Initial local REMB bandwidth estimation that gets used when a new endpoint is connected. Only useful for connections that are compatible with the REMB bandwidth estimation protocol (such as most WebRTC peers).

All bandwidth control parameters must be set before the SDP negotiation takes place, and can't be modified afterwards.

Take into consideration that setting a too high upper limit for the output bandwidth can be a reason for the network connection to be congested quickly.

Constructor

(abstract) new BaseRtpEndpoint()

Source:
Fires:

Extends

Members

(static) constructorParams

Source:

(static) events

Source:

Methods

(static) requestKeyframe(callbackopt) → {external:Promise}

Force sending a new keyframe request to the upstream elements in the Kurento Pipeline, towards the associated producer. Only valid for video consumers.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~requestKeyframeCallback <optional>
Source:
Returns:
Type
external:Promise

getChildren(callbackopt) → {external:Promise}

Children of this MediaObject.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getChildrenCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getConnectionState(callbackopt) → {external:Promise}

Connection state.
  • CONNECTED
  • DISCONNECTED
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getConnectionStateCallback <optional>
Source:
Returns:
Type
external:Promise

getCreationTime(callbackopt) → {external:Promise}

MediaObject creation time in seconds since Epoch.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getCreationTimeCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getEncoderBitrate(callbackopt) → {external:Promise}

Target video bitrate for media transcoding.

The bitrate of a video has a direct impact on its perceived image quality. Higher bitrate means higher quality, but also a larger amount of bytes to transmit or store. Use this parameter to set the desired average bitrate in videos that are transcoded by the media server.

This parameter is most useful for :rom:cls:`RecorderEndpoint` and :rom:cls:`RtpEndpoint`: when media is being transcoded (either for streaming or storing on disk), the resulting quality is directly controlled with this value.

For :rom:cls:`WebRtcEndpoint`, this value should be left as default, as remote WebRTC receivers will already send feedback to inform the media server about what is the optimal bitrate to send.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 300000 (300 kbps).
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxAudioRecvBandwidth(callbackopt) → {external:Promise}

Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
  • Unit: kbps (kilobits per second).
  • Default: 0 (unlimited).
Parameters:
Name Type Attributes Description
callback module:core/abstracts.SdpEndpoint~getMaxAudioRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxEncoderBitrate(callbackopt) → {external:Promise}

Maximum video bitrate for media transcoding.

This parameter can be used to fine tune the automatic bitrate selection that normally takes place within elements that are able to dynamically change the encoding bitrate according to the conditions of the streaming, such as :rom:cls:`WebRtcEndpoint`.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to limit the total bitrate that is handled by the server, by setting a low maximum output for all endpoints.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to limit the total bitrate that is handled by the server, by setting a low maximum output for all endpoints.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 0.
  • 0 = unlimited. Encoding performed with bitrate as requested by receivers.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getMaxEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxVideoRecvBandwidth(callbackopt) → {external:Promise}

Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
  • Unit: kbps (kilobits per second).
  • Default: 0 (unlimited).
Parameters:
Name Type Attributes Description
callback module:core/abstracts.SdpEndpoint~getMaxVideoRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxVideoSendBandwidth(callbackopt) → {external:Promise}

Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

With this parameter you can control the maximum video quality that will be sent when reacting to good network conditions. Setting this parameter to a high value permits the video quality to raise when the network conditions get better.

Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

The REMB congestion control algorithm works by gradually increasing the output video bitrate, until the available bandwidth is fully used or the maximum send bitrate has been reached. This is a slow, progressive change, which starts at 300 kbps by default. You can change the default starting point of REMB estimations, by setting RembParams.rembOnConnect.

  • Unit: kbps (kilobits per second).
  • Default: 500.
  • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
    Note that this might have a bad effect if more than one stream is running (as all of them would try to raise the video bitrate indefinitely, until the network gets saturated).
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getMaxVideoSendBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

getMediaPipeline(callbackopt) → {external:Promise}

MediaPipeline to which this MediaObject belongs. It returns itself when invoked for a pipeline object.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getMediaPipelineCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMediaState(callbackopt) → {external:Promise}

Media flow state.
  • CONNECTED: There is an RTCP flow.
  • DISCONNECTED: No RTCP packets have been received for at least 5 sec.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getMediaStateCallback <optional>
Source:
Returns:
Type
external:Promise

getMinEncoderBitrate(callbackopt) → {external:Promise}

Minimum video bitrate for media transcoding.

This parameter can be used to fine tune the automatic bitrate selection that normally takes place within elements that are able to dynamically change the encoding bitrate according to the conditions of the streaming, such as :rom:cls:`WebRtcEndpoint`.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to force a higher bitrate than what is being requested by receivers.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 0.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getMinEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMinVideoRecvBandwidth(callbackopt) → {external:Promise}

Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • Note: The actual minimum value is 30 kbps, even if a lower value is set.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getMinVideoRecvBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

getMinVideoSendBandwidth(callbackopt) → {external:Promise}

Minimum outbound bitrate sent by this endpoint.

With this parameter you can control the minimum video quality that will be sent when reacting to bad network conditions. Setting this parameter to a low value permits the video quality to drop when the network conditions get worse.

Note that if you set this parameter too high (trying to avoid bad video quality altogether), you would be limiting the adaptation ability of the congestion control algorithm, and your stream might be unable to ever recover from adverse network conditions.

  • Unit: kbps (kilobits per second).
  • Default: 100.
  • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getMinVideoSendBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

getMtu(callbackopt) → {external:Promise}

Maximum Transmission Unit (MTU) used for RTP.

This setting affects the maximum size that will be used by RTP payloads. You can change it from the default, if you think that a different value would be beneficial for the typical network settings of your application.

The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

WARNING: Change this value ONLY if you really know what you are doing and you have strong reasons to do so. Do NOT change this parameter just because it seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

  • Unit: Bytes.
  • Default: 1200.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getMtuCallback <optional>
Source:
Returns:
Type
external:Promise

getName(callbackopt) → {external:Promise}

This MediaObject's name.

This is just sugar to simplify developers' life debugging, it is not used internally for indexing nor identifying the objects. By default, it's the object's ID.

Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getNameCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getParent(callbackopt) → {external:Promise}

Parent of this MediaObject.

The parent of a Hub or a MediaElement is its MediaPipeline. A MediaPipeline has no parent, so this property will be null.

Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getParentCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getRembParams(callbackopt) → {external:Promise}

Advanced parameters to configure the congestion control algorithm.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.BaseRtpEndpoint~getRembParamsCallback <optional>
Source:
Returns:
Type
external:Promise

getSendTagsInEvents(callbackopt) → {external:Promise}

Flag activating or deactivating sending the element's tags in fired events.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getSendTagsInEventsCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setEncoderBitrate(encoderBitrate, callbackopt) → {external:Promise}

Target video bitrate for media transcoding.

The bitrate of a video has a direct impact on its perceived image quality. Higher bitrate means higher quality, but also a larger amount of bytes to transmit or store. Use this parameter to set the desired average bitrate in videos that are transcoded by the media server.

This parameter is most useful for :rom:cls:`RecorderEndpoint` and :rom:cls:`RtpEndpoint`: when media is being transcoded (either for streaming or storing on disk), the resulting quality is directly controlled with this value.

For :rom:cls:`WebRtcEndpoint`, this value should be left as default, as remote WebRTC receivers will already send feedback to inform the media server about what is the optimal bitrate to send.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 300000 (300 kbps).
Parameters:
Name Type Attributes Description
encoderBitrate external:Integer
callback module:core/abstracts.MediaElement~setEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMaxAudioRecvBandwidth(maxAudioRecvBandwidth, callbackopt) → {external:Promise}

Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
  • Unit: kbps (kilobits per second).
  • Default: 0 (unlimited).
Parameters:
Name Type Attributes Description
maxAudioRecvBandwidth external:Integer
callback module:core/abstracts.SdpEndpoint~setMaxAudioRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMaxEncoderBitrate(maxEncoderBitrate, callbackopt) → {external:Promise}

Maximum video bitrate for media transcoding.

This parameter can be used to fine tune the automatic bitrate selection that normally takes place within elements that are able to dynamically change the encoding bitrate according to the conditions of the streaming, such as :rom:cls:`WebRtcEndpoint`.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to limit the total bitrate that is handled by the server, by setting a low maximum output for all endpoints.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to limit the total bitrate that is handled by the server, by setting a low maximum output for all endpoints.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 0.
  • 0 = unlimited. Encoding performed with bitrate as requested by receivers.
Parameters:
Name Type Attributes Description
maxEncoderBitrate external:Integer
callback module:core/abstracts.MediaElement~setMaxEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMaxVideoRecvBandwidth(maxVideoRecvBandwidth, callbackopt) → {external:Promise}

Maximum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders as part of Kurento's REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers.
  • Unit: kbps (kilobits per second).
  • Default: 0 (unlimited).
Parameters:
Name Type Attributes Description
maxVideoRecvBandwidth external:Integer
callback module:core/abstracts.SdpEndpoint~setMaxVideoRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMaxVideoSendBandwidth(maxVideoSendBandwidth, callbackopt) → {external:Promise}

Maximum outbound bitrate sent by this endpoint. Remote peers themselves might also announce a maximum bitrate as part of their REMB bandwidth estimations, and also as an SDP bitrate attribute (b=AS:{value}, see RFC 8866 Section 5.8. Bandwidth Information) in sendrecv and recvonly SDP Offers or Answers. Kurento will always give priority to bitrate limitations specified by remote peers, over internal configurations.

With this parameter you can control the maximum video quality that will be sent when reacting to good network conditions. Setting this parameter to a high value permits the video quality to raise when the network conditions get better.

Note that the default MaxVideoSendBandwidth is a VERY conservative value, and leads to a low maximum video quality. Most applications will probably want to increase this to higher values such as 2000 kbps (2 Mbps).

The REMB congestion control algorithm works by gradually increasing the output video bitrate, until the available bandwidth is fully used or the maximum send bitrate has been reached. This is a slow, progressive change, which starts at 300 kbps by default. You can change the default starting point of REMB estimations, by setting RembParams.rembOnConnect.

  • Unit: kbps (kilobits per second).
  • Default: 500.
  • 0 = unlimited: the video bitrate will grow until all the available network bandwidth is used by the stream.
    Note that this might have a bad effect if more than one stream is running (as all of them would try to raise the video bitrate indefinitely, until the network gets saturated).
Parameters:
Name Type Attributes Description
maxVideoSendBandwidth external:Integer
callback module:core/abstracts.BaseRtpEndpoint~setMaxVideoSendBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

setMinEncoderBitrate(minEncoderBitrate, callbackopt) → {external:Promise}

Minimum video bitrate for media transcoding.

This parameter can be used to fine tune the automatic bitrate selection that normally takes place within elements that are able to dynamically change the encoding bitrate according to the conditions of the streaming, such as :rom:cls:`WebRtcEndpoint`.

This should be left as default in most cases, given that remote WebRTC receivers already send feedback to inform the media server about what is the optimal bitrate to send. Otherwise, this parameter could be used for example to force a higher bitrate than what is being requested by receivers.

Setting a value will only work if done before the media starts to flow.

  • Unit: bps (bits per second).
  • Default: 0.
Parameters:
Name Type Attributes Description
minEncoderBitrate external:Integer
callback module:core/abstracts.MediaElement~setMinEncoderBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMinVideoRecvBandwidth(minVideoRecvBandwidth, callbackopt) → {external:Promise}

Minimum inbound bitrate requested by this endpoint. Signaled to compatible WebRTC and RTP senders, as part of the REMB bandwidth estimation protocol.

This is used to set a minimum value of local REMB during bandwidth estimation. The REMB estimation will then be sent to remote peers, causing them to send at least the indicated video bitrate. It follows that this parameter will only have effect for remote peers that support the REMB bandwidth estimation protocol (such as e.g. most web browsers compatible with WebRTC).

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • Note: The actual minimum value is 30 kbps, even if a lower value is set.
Parameters:
Name Type Attributes Description
minVideoRecvBandwidth external:Integer
callback module:core/abstracts.BaseRtpEndpoint~setMinVideoRecvBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

setMinVideoSendBandwidth(minVideoSendBandwidth, callbackopt) → {external:Promise}

Minimum outbound bitrate sent by this endpoint.

With this parameter you can control the minimum video quality that will be sent when reacting to bad network conditions. Setting this parameter to a low value permits the video quality to drop when the network conditions get worse.

Note that if you set this parameter too high (trying to avoid bad video quality altogether), you would be limiting the adaptation ability of the congestion control algorithm, and your stream might be unable to ever recover from adverse network conditions.

  • Unit: kbps (kilobits per second).
  • Default: 100.
  • 0 = unlimited: the video bitrate will drop as needed, even to the lowest possible quality, which could make the video completely blurry and pixelated, but would adapt better to losses caused by network congestion.
Parameters:
Name Type Attributes Description
minVideoSendBandwidth external:Integer
callback module:core/abstracts.BaseRtpEndpoint~setMinVideoSendBandwidthCallback <optional>
Source:
Returns:
Type
external:Promise

setMtu(mtu, callbackopt) → {external:Promise}

Maximum Transmission Unit (MTU) used for RTP.

This setting affects the maximum size that will be used by RTP payloads. You can change it from the default, if you think that a different value would be beneficial for the typical network settings of your application.

The default value is 1200 Bytes. This is the same as in libwebrtc (from webrtc.org), as used by Firefox or Chrome . You can read more about this value in Why RTP max packet size is 1200 in WebRTC? .

WARNING: Change this value ONLY if you really know what you are doing and you have strong reasons to do so. Do NOT change this parameter just because it seems to work better for some reduced scope tests. The default value is a consensus chosen by people who have deep knowledge about network optimization.

  • Unit: Bytes.
  • Default: 1200.
Parameters:
Name Type Attributes Description
mtu external:Integer
callback module:core/abstracts.BaseRtpEndpoint~setMtuCallback <optional>
Source:
Returns:
Type
external:Promise

setName(name, callbackopt) → {external:Promise}

This MediaObject's name.

This is just sugar to simplify developers' life debugging, it is not used internally for indexing nor identifying the objects. By default, it's the object's ID.

Parameters:
Name Type Attributes Description
name external:String
callback module:core/abstracts.MediaObject~setNameCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setRembParams(rembParams, callbackopt) → {external:Promise}

Advanced parameters to configure the congestion control algorithm.
Parameters:
Name Type Attributes Description
rembParams module:core/complexTypes.RembParams
callback module:core/abstracts.BaseRtpEndpoint~setRembParamsCallback <optional>
Source:
Returns:
Type
external:Promise

setSendTagsInEvents(sendTagsInEvents, callbackopt) → {external:Promise}

Flag activating or deactivating sending the element's tags in fired events.
Parameters:
Name Type Attributes Description
sendTagsInEvents external:Boolean
callback module:core/abstracts.MediaObject~setSendTagsInEventsCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

Type Definitions

getConnectionStateCallback(error, result)

Parameters:
Name Type Description
error external:Error
result module:core/complexTypes.ConnectionState
Source:

getMaxVideoSendBandwidthCallback(error, result)

Parameters:
Name Type Description
error external:Error
result external:Integer
Source:

getMediaStateCallback(error, result)

Parameters:
Name Type Description
error external:Error
result module:core/complexTypes.MediaState
Source:

getMinVideoRecvBandwidthCallback(error, result)

Parameters:
Name Type Description
error external:Error
result external:Integer
Source:

getMinVideoSendBandwidthCallback(error, result)

Parameters:
Name Type Description
error external:Error
result external:Integer
Source:

getMtuCallback(error, result)

Parameters:
Name Type Description
error external:Error
result external:Integer
Source:

getRembParamsCallback(error, result)

Parameters:
Name Type Description
error external:Error
result module:core/complexTypes.RembParams
Source:

requestKeyframeCallback(error)

Parameters:
Name Type Description
error external:Error
Source:

setMaxVideoSendBandwidthCallback(error)

Parameters:
Name Type Description
error external:Error
Source:

setMinVideoRecvBandwidthCallback(error)

Parameters:
Name Type Description
error external:Error
Source:

setMinVideoSendBandwidthCallback(error)

Parameters:
Name Type Description
error external:Error
Source:

setMtuCallback(error)

Parameters:
Name Type Description
error external:Error
Source:

setRembParamsCallback(error)

Parameters:
Name Type Description
error external:Error
Source: