Class: BaseRtpEndpoint

(abstract) core/abstracts.BaseRtpEndpoint()

Handles RTP communications.

All endpoints that rely on the RTP protocol, like the RtpEndpoint or the WebRtcEndpoint, inherit from this class. The endpoint provides information about the connection state and the media state, which can be consulted at any time through the module:core/abstracts.BaseRtpEndpoint#mediaState and the module:core/abstracts.BaseRtpEndpoint#connectionState properties. 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.

    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 module:core/abstracts.BaseRtpEndpoint#MediaFlowInStateChange and module:core/abstracts.BaseRtpEndpoint#MediaFlowOutStateChange, 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.

    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.

Part of the bandwidth control for the video component of the media session done here:

  • Input bandwidth: Values used to inform remote peers about the bitrate that can be sent to this endpoint.
    • MinVideoRecvBandwidth: Minimum input bitrate, requested from WebRTC senders with REMB (Default: 30 Kbps).
    • MaxAudioRecvBandwidth and MaxVideoRecvBandwidth: Maximum input bitrate, signaled in SDP Offers to WebRTC and RTP senders (Default: unlimited).
  • Output bandwidth: Values used to control bitrate of the video streams sent to remote peers. It is important to keep in mind that pushed bitrate depends on network and remote peer capabilities. Remote peers can also announce bandwidth limitation in their SDPs (through the b={modifier}:{value} attribute). Kurento will always enforce bitrate limitations specified by the remote peer over internal configurations.
    • MinVideoSendBandwidth: REMB override of minimum bitrate sent to WebRTC receivers (Default: 100 Kbps).
    • MaxVideoSendBandwidth: REMB override of maximum bitrate sent to WebRTC receivers (Default: 500 Kbps).
    • RembParams.rembOnConnect: Initial local REMB bandwidth estimation that gets propagated when a new endpoint is connected.

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

Constructor

(abstract) new BaseRtpEndpoint()

Source:
Fires:

Extends

Members

(static) constructorParams

Source:

(static) events

Source:

Methods

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

getChilds(callbackopt) → {external:Promise}

Children of this MediaObject.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaObject~getChildsCallback <optional>
Inherited From:
Deprecated:
  • Use children instead.
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

getMaxAudioRecvBandwidth(callbackopt) → {external:Promise}

Maximum input bitrate, signaled in SDP Offers to WebRTC and RTP senders.

This is used to put a limit on the bitrate that the remote peer will send to this endpoint. The net effect of setting this parameter is that when Kurento generates an SDP Offer, an 'Application Specific' (AS) maximum bandwidth attribute will be added to the SDP media section: b=AS:{value}.

Note: This parameter has to be set before the SDP is generated.

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • 0 = unlimited.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.SdpEndpoint~getMaxAudioRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxOuputBitrate(callbackopt) → {external:Promise}

Maximum video bandwidth for transcoding.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getMaxOuputBitrateCallback <optional>
Inherited From:
Deprecated:
  • Deprecated due to a typo. Use module:core/abstracts.MediaElement#maxOutputBitrate instead of this function.
Source:
Returns:
Type
external:Promise

getMaxOutputBitrate(callbackopt) → {external:Promise}

Maximum video bitrate for transcoding.
  • Unit: bps (bits per second).
  • Default: MAXINT.
  • 0 = unlimited.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getMaxOutputBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

getMaxVideoRecvBandwidth(callbackopt) → {external:Promise}

Maximum input bitrate, signaled in SDP Offers to WebRTC and RTP senders.

This is used to put a limit on the bitrate that the remote peer will send to this endpoint. The net effect of setting this parameter is that when Kurento generates an SDP Offer, an 'Application Specific' (AS) maximum bandwidth attribute will be added to the SDP media section: b=AS:{value}.

Note: This parameter has to be set before the SDP is generated.

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • 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}

REMB override of maximum bitrate sent to WebRTC receivers.

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.

This parameter provides a way to limit the bitrate requested by remote REMB bandwidth estimations: the bitrate sent will be always equal or less than this parameter, even if the remote peer requests higher bitrates.

Note that the default value of 500 kbps is a VERY conservative one, 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

getMinOuputBitrate(callbackopt) → {external:Promise}

Minimum video bandwidth for transcoding.
Parameters:
Name Type Attributes Description
callback module:core/abstracts.MediaElement~getMinOuputBitrateCallback <optional>
Inherited From:
Deprecated:
  • Deprecated due to a typo. Use module:core/abstracts.MediaElement#minOutputBitrate instead of this function.
Source:
Returns:
Type
external:Promise

getMinOutputBitrate(callbackopt) → {external:Promise}

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

getMinVideoRecvBandwidth(callbackopt) → {external:Promise}

Minimum input bitrate, requested from WebRTC senders with REMB.

This is used to set a minimum value of local REMB during bandwidth estimation, if supported by the implementing class. The REMB estimation will then be sent to remote peers, requesting them to send at least the indicated video bitrate. It follows that min values will only have effect in remote peers that support this congestion control mechanism, such as Chrome.

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

getMinVideoSendBandwidth(callbackopt) → {external:Promise}

REMB override of minimum bitrate sent to WebRTC receivers.

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.

This parameter provides a way to override the bitrate requested by remote REMB bandwidth estimations: the bitrate sent will be always equal or greater than this parameter, even if the remote peer requests even lower bitrates.

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 might make the video completely blurry and pixelated.
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

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

Maximum input bitrate, signaled in SDP Offers to WebRTC and RTP senders.

This is used to put a limit on the bitrate that the remote peer will send to this endpoint. The net effect of setting this parameter is that when Kurento generates an SDP Offer, an 'Application Specific' (AS) maximum bandwidth attribute will be added to the SDP media section: b=AS:{value}.

Note: This parameter has to be set before the SDP is generated.

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • 0 = unlimited.
Parameters:
Name Type Attributes Description
maxAudioRecvBandwidth external:Integer
callback module:core/abstracts.SdpEndpoint~setMaxAudioRecvBandwidthCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

setMaxOuputBitrate(maxOuputBitrate, callbackopt) → {external:Promise}

Maximum video bandwidth for transcoding.
Parameters:
Name Type Attributes Description
maxOuputBitrate external:Integer
callback module:core/abstracts.MediaElement~setMaxOuputBitrateCallback <optional>
Inherited From:
Deprecated:
  • Deprecated due to a typo. Use module:core/abstracts.MediaElement#maxOutputBitrate instead of this function.
Source:
Returns:
Type
external:Promise

setMaxOutputBitrate(maxOutputBitrate, callbackopt) → {external:Promise}

Maximum video bitrate for transcoding.
  • Unit: bps (bits per second).
  • Default: MAXINT.
  • 0 = unlimited.
Parameters:
Name Type Attributes Description
maxOutputBitrate external:Integer
callback module:core/abstracts.MediaElement~setMaxOutputBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

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

Maximum input bitrate, signaled in SDP Offers to WebRTC and RTP senders.

This is used to put a limit on the bitrate that the remote peer will send to this endpoint. The net effect of setting this parameter is that when Kurento generates an SDP Offer, an 'Application Specific' (AS) maximum bandwidth attribute will be added to the SDP media section: b=AS:{value}.

Note: This parameter has to be set before the SDP is generated.

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • 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}

REMB override of maximum bitrate sent to WebRTC receivers.

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.

This parameter provides a way to limit the bitrate requested by remote REMB bandwidth estimations: the bitrate sent will be always equal or less than this parameter, even if the remote peer requests higher bitrates.

Note that the default value of 500 kbps is a VERY conservative one, 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

setMinOuputBitrate(minOuputBitrate, callbackopt) → {external:Promise}

Minimum video bandwidth for transcoding.
Parameters:
Name Type Attributes Description
minOuputBitrate external:Integer
callback module:core/abstracts.MediaElement~setMinOuputBitrateCallback <optional>
Inherited From:
Deprecated:
  • Deprecated due to a typo. Use module:core/abstracts.MediaElement#minOutputBitrate instead of this function.
Source:
Returns:
Type
external:Promise

setMinOutputBitrate(minOutputBitrate, callbackopt) → {external:Promise}

Minimum video bitrate for transcoding.
  • Unit: bps (bits per second).
  • Default: 0.
Parameters:
Name Type Attributes Description
minOutputBitrate external:Integer
callback module:core/abstracts.MediaElement~setMinOutputBitrateCallback <optional>
Inherited From:
Source:
Returns:
Type
external:Promise

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

Minimum input bitrate, requested from WebRTC senders with REMB.

This is used to set a minimum value of local REMB during bandwidth estimation, if supported by the implementing class. The REMB estimation will then be sent to remote peers, requesting them to send at least the indicated video bitrate. It follows that min values will only have effect in remote peers that support this congestion control mechanism, such as Chrome.

  • Unit: kbps (kilobits per second).
  • Default: 0.
  • Note: The absolute minimum REMB value is 30 kbps, even if a lower value is set here.
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}

REMB override of minimum bitrate sent to WebRTC receivers.

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.

This parameter provides a way to override the bitrate requested by remote REMB bandwidth estimations: the bitrate sent will be always equal or greater than this parameter, even if the remote peer requests even lower bitrates.

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 might make the video completely blurry and pixelated.
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:

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: