Class: RtpEndpoint

elements.RtpEndpoint()

Endpoint that provides bidirectional content delivery capabilities with remote networked peers through RTP or SRTP protocol. An RtpEndpoint contains paired sink and source :rom:cls:`MediaPad` for audio and video. This endpoint inherits from BaseRtpEndpoint.

In order to establish an RTP/SRTP communication, peers engage in an SDP negotiation process, where one of the peers (the offerer) sends an

  • As offerer: The negotiation process is initiated by the media server
    • KMS generates the SDP offer through the generateOffer method. This offer must then be sent to the remote peer (the offeree) through the signaling channel, for processing.
    • The remote peer process the Offer, and generates an Answer to this offer. The Answer is sent back to the media server.
    • Upon receiving the Answer, the endpoint must invoke the processAnswer method.
  • As offeree: The negotiation process is initiated by the remote peer
    • The remote peer, acting as offerer, generates an SDP offer and sends it to the WebRTC endpoint in Kurento.
    • The endpoint will process the Offer invoking the processOffer method. The result of this method will be a string,
    • The SDP Answer must be sent back to the offerer, so it can be processed.

In case of unidirectional connections (i.e. only one peer is going to send media), the process is more simple, as only the emitter needs to process an SDP. On top of the information about media codecs and types, the SDP must contain the IP of the remote peer, and the port where it will be listening. This way, the SDP can be mangled without needing to go through the exchange process, as the receiving peer does

The user can set some bandwidth limits that will be used during the negotiation process. The default bandwidth range of the endpoint is 100kbps-500kbps, but it

  • Input bandwidth control mechanism: Configuration interval used to inform remote peer the range of bitrates that can be pushed into this RtpEndpoint object. These values are announced in the SDP.
    • setMaxVideoRecvBandwidth: sets Max bitrate limits expected for
    • setMaxAudioRecvBandwidth: sets Max bitrate limits expected for
  • Output bandwidth control mechanism: Configuration interval used to
    • setMaxVideoSendBandwidth: sets Max bitrate limits for video sent to remote peer.
    • setMinVideoSendBandwidth: sets Min bitrate limits for audio sent to remote peer.
All bandwidth control parameters must be changed before the SDP negotiation takes place, and can't be modified afterwards. TODO: What happens if the b=as tag form the SDP has a lower value than

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

Constructor

new RtpEndpoint()

Builder for the RtpEndpoint
Source:
Fires:

Extends

Members

(static) constructorParams

Properties:
Name Type Attributes Description
crypto module:elements/complexTypes.SDES <optional>
SDES-type param. If present, this parameter indicates that the communication
mediaPipeline module:core.MediaPipeline the MediaPipeline to which the endpoint belongs
useIpv6 external:Boolean <optional>
This configures the endpoint to use IPv6 instead of IPv4.
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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>
Inherited From:
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