org.kurento.client

Interface BaseRtpEndpoint

All Superinterfaces:

Endpoint, KurentoObject, MediaElement, MediaObject, SdpEndpoint, SessionEndpoint

All Known Subinterfaces:

RtpEndpoint, WebRtcEndpoint

public interface BaseRtpEndpoint
extends SdpEndpoint

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 getMediaState() and the 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 any 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 the 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.

Part of the bandwidth control for the video component of the media session is 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.

Method Summary

Modifier and Type	Method and Description
ListenerSubscription	addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener) Add a EventListener for event ConnectionStateChangedEvent.
void	addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener, Continuation<ListenerSubscription> cont) Add a EventListener for event ConnectionStateChangedEvent.
ListenerSubscription	addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener) Add a EventListener for event MediaStateChangedEvent.
void	addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener, Continuation<ListenerSubscription> cont) Add a EventListener for event MediaStateChangedEvent.
ConnectionState	getConnectionState() Get Connection state.
void	getConnectionState(Continuation<ConnectionState> cont) Get Connection state.
TFuture<ConnectionState>	getConnectionState(Transaction tx) Get Connection state.
int	getMaxVideoSendBandwidth() Get REMB override of maximum bitrate sent to WebRTC receivers.
void	getMaxVideoSendBandwidth(Continuation<Integer> cont) Get REMB override of maximum bitrate sent to WebRTC receivers.
TFuture<Integer>	getMaxVideoSendBandwidth(Transaction tx) Get REMB override of maximum bitrate sent to WebRTC receivers.
MediaState	getMediaState() Get Media flow state.
void	getMediaState(Continuation<MediaState> cont) Get Media flow state.
TFuture<MediaState>	getMediaState(Transaction tx) Get Media flow state.
int	getMinVideoRecvBandwidth() Get Minimum input bitrate, requested from WebRTC senders with REMB.
void	getMinVideoRecvBandwidth(Continuation<Integer> cont) Get Minimum input bitrate, requested from WebRTC senders with REMB.
TFuture<Integer>	getMinVideoRecvBandwidth(Transaction tx) Get Minimum input bitrate, requested from WebRTC senders with REMB.
int	getMinVideoSendBandwidth() Get REMB override of minimum bitrate sent to WebRTC receivers.
void	getMinVideoSendBandwidth(Continuation<Integer> cont) Get REMB override of minimum bitrate sent to WebRTC receivers.
TFuture<Integer>	getMinVideoSendBandwidth(Transaction tx) Get REMB override of minimum bitrate sent to WebRTC receivers.
int	getMtu() Get Maximum Transmission Unit (MTU) used for RTP.
void	getMtu(Continuation<Integer> cont) Get Maximum Transmission Unit (MTU) used for RTP.
TFuture<Integer>	getMtu(Transaction tx) Get Maximum Transmission Unit (MTU) used for RTP.
RembParams	getRembParams() Get Advanced parameters to configure the congestion control algorithm.
void	getRembParams(Continuation<RembParams> cont) Get Advanced parameters to configure the congestion control algorithm.
TFuture<RembParams>	getRembParams(Transaction tx) Get Advanced parameters to configure the congestion control algorithm.
void	removeConnectionStateChangedListener(ListenerSubscription listenerSubscription) Remove a ListenerSubscription for event ConnectionStateChangedEvent.
void	removeConnectionStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont) Remove a ListenerSubscription for event ConnectionStateChangedEvent.
void	removeMediaStateChangedListener(ListenerSubscription listenerSubscription) Remove a ListenerSubscription for event MediaStateChangedEvent.
void	removeMediaStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont) Remove a ListenerSubscription for event MediaStateChangedEvent.
void	setMaxVideoSendBandwidth(int maxVideoSendBandwidth) Set REMB override of maximum bitrate sent to WebRTC receivers.
void	setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Continuation<Void> cont) Set REMB override of maximum bitrate sent to WebRTC receivers.
void	setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Transaction tx) Set REMB override of maximum bitrate sent to WebRTC receivers.
void	setMinVideoRecvBandwidth(int minVideoRecvBandwidth) Set Minimum input bitrate, requested from WebRTC senders with REMB.
void	setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Continuation<Void> cont) Set Minimum input bitrate, requested from WebRTC senders with REMB.
void	setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Transaction tx) Set Minimum input bitrate, requested from WebRTC senders with REMB.
void	setMinVideoSendBandwidth(int minVideoSendBandwidth) Set REMB override of minimum bitrate sent to WebRTC receivers.
void	setMinVideoSendBandwidth(int minVideoSendBandwidth, Continuation<Void> cont) Set REMB override of minimum bitrate sent to WebRTC receivers.
void	setMinVideoSendBandwidth(int minVideoSendBandwidth, Transaction tx) Set REMB override of minimum bitrate sent to WebRTC receivers.
void	setMtu(int mtu) Set Maximum Transmission Unit (MTU) used for RTP.
void	setMtu(int mtu, Continuation<Void> cont) Set Maximum Transmission Unit (MTU) used for RTP.
void	setMtu(int mtu, Transaction tx) Set Maximum Transmission Unit (MTU) used for RTP.
void	setRembParams(RembParams rembParams) Set Advanced parameters to configure the congestion control algorithm.
void	setRembParams(RembParams rembParams, Continuation<Void> cont) Set Advanced parameters to configure the congestion control algorithm.
void	setRembParams(RembParams rembParams, Transaction tx) Set Advanced parameters to configure the congestion control algorithm.

Methods inherited from interface org.kurento.client.SdpEndpoint

generateOffer, generateOffer, generateOffer, generateOffer, generateOffer, generateOffer, getLocalSessionDescriptor, getLocalSessionDescriptor, getLocalSessionDescriptor, getMaxAudioRecvBandwidth, getMaxAudioRecvBandwidth, getMaxAudioRecvBandwidth, getMaxVideoRecvBandwidth, getMaxVideoRecvBandwidth, getMaxVideoRecvBandwidth, getRemoteSessionDescriptor, getRemoteSessionDescriptor, getRemoteSessionDescriptor, processAnswer, processAnswer, processAnswer, processOffer, processOffer, processOffer, setMaxAudioRecvBandwidth, setMaxAudioRecvBandwidth, setMaxAudioRecvBandwidth, setMaxVideoRecvBandwidth, setMaxVideoRecvBandwidth, setMaxVideoRecvBandwidth

Methods inherited from interface org.kurento.client.SessionEndpoint

addMediaSessionStartedListener, addMediaSessionStartedListener, addMediaSessionTerminatedListener, addMediaSessionTerminatedListener, removeMediaSessionStartedListener, removeMediaSessionStartedListener, removeMediaSessionTerminatedListener, removeMediaSessionTerminatedListener

Methods inherited from interface org.kurento.client.MediaElement

addElementConnectedListener, addElementConnectedListener, addElementDisconnectedListener, addElementDisconnectedListener, addMediaFlowInStateChangedListener, addMediaFlowInStateChangedListener, addMediaFlowInStateChangeListener, addMediaFlowInStateChangeListener, addMediaFlowOutStateChangedListener, addMediaFlowOutStateChangedListener, addMediaFlowOutStateChangeListener, addMediaFlowOutStateChangeListener, addMediaTranscodingStateChangedListener, addMediaTranscodingStateChangedListener, addMediaTranscodingStateChangeListener, addMediaTranscodingStateChangeListener, connect, connect, connect, connect, connect, connect, connect, connect, connect, connect, connect, connect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, disconnect, getGstreamerDot, getGstreamerDot, getGstreamerDot, getGstreamerDot, getGstreamerDot, getGstreamerDot, getMaxOuputBitrate, getMaxOuputBitrate, getMaxOuputBitrate, getMaxOutputBitrate, getMaxOutputBitrate, getMaxOutputBitrate, getMinOuputBitrate, getMinOuputBitrate, getMinOuputBitrate, getMinOutputBitrate, getMinOutputBitrate, getMinOutputBitrate, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSinkConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getSourceConnections, getStats, getStats, getStats, getStats, getStats, getStats, isMediaFlowingIn, isMediaFlowingIn, isMediaFlowingIn, isMediaFlowingIn, isMediaFlowingIn, isMediaFlowingIn, isMediaFlowingOut, isMediaFlowingOut, isMediaFlowingOut, isMediaFlowingOut, isMediaFlowingOut, isMediaFlowingOut, isMediaTranscoding, isMediaTranscoding, isMediaTranscoding, isMediaTranscoding, isMediaTranscoding, isMediaTranscoding, removeElementConnectedListener, removeElementConnectedListener, removeElementDisconnectedListener, removeElementDisconnectedListener, removeMediaFlowInStateChangedListener, removeMediaFlowInStateChangedListener, removeMediaFlowInStateChangeListener, removeMediaFlowInStateChangeListener, removeMediaFlowOutStateChangedListener, removeMediaFlowOutStateChangedListener, removeMediaFlowOutStateChangeListener, removeMediaFlowOutStateChangeListener, removeMediaTranscodingStateChangedListener, removeMediaTranscodingStateChangedListener, removeMediaTranscodingStateChangeListener, removeMediaTranscodingStateChangeListener, setAudioFormat, setAudioFormat, setAudioFormat, setMaxOuputBitrate, setMaxOuputBitrate, setMaxOuputBitrate, setMaxOutputBitrate, setMaxOutputBitrate, setMaxOutputBitrate, setMinOuputBitrate, setMinOuputBitrate, setMinOuputBitrate, setMinOutputBitrate, setMinOutputBitrate, setMinOutputBitrate, setOutputBitrate, setOutputBitrate, setOutputBitrate, setVideoFormat, setVideoFormat, setVideoFormat

Methods inherited from interface org.kurento.client.MediaObject

addErrorListener, addErrorListener, addTag, addTag, addTag, getChildren, getChildren, getChildren, getChilds, getChilds, getChilds, getCreationTime, getCreationTime, getCreationTime, getId, getId, getId, getMediaPipeline, getMediaPipeline, getMediaPipeline, getName, getName, getName, getParent, getParent, getParent, getSendTagsInEvents, getSendTagsInEvents, getSendTagsInEvents, getTag, getTag, getTag, getTags, getTags, getTags, removeErrorListener, removeErrorListener, removeTag, removeTag, removeTag, setName, setName, setName, setSendTagsInEvents, setSendTagsInEvents, setSendTagsInEvents

Methods inherited from interface org.kurento.client.KurentoObject

addEventListener, invoke, isCommited, release, release, release, removeEventListener, waitCommited, whenCommited, whenCommited

Method Detail

getMinVideoRecvBandwidth

int getMinVideoRecvBandwidth()

Get 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.

getMinVideoRecvBandwidth

void getMinVideoRecvBandwidth(Continuation<Integer> cont)

Get 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.

getMinVideoRecvBandwidth

TFuture<Integer> getMinVideoRecvBandwidth(Transaction tx)

Get 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.

setMinVideoRecvBandwidth

void setMinVideoRecvBandwidth(int minVideoRecvBandwidth)

Set 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.

setMinVideoRecvBandwidth

void setMinVideoRecvBandwidth(int minVideoRecvBandwidth,
                              Continuation<Void> cont)

Set 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.

setMinVideoRecvBandwidth

void setMinVideoRecvBandwidth(int minVideoRecvBandwidth,
                              Transaction tx)

Set 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.

getMinVideoSendBandwidth

int getMinVideoSendBandwidth()

Get 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.

getMinVideoSendBandwidth

void getMinVideoSendBandwidth(Continuation<Integer> cont)

Get 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.

getMinVideoSendBandwidth

TFuture<Integer> getMinVideoSendBandwidth(Transaction tx)

Get 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.

setMinVideoSendBandwidth

void setMinVideoSendBandwidth(int minVideoSendBandwidth)

Set 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.

setMinVideoSendBandwidth

void setMinVideoSendBandwidth(int minVideoSendBandwidth,
                              Continuation<Void> cont)

Set 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.

setMinVideoSendBandwidth

void setMinVideoSendBandwidth(int minVideoSendBandwidth,
                              Transaction tx)

Set 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.

getMaxVideoSendBandwidth

int getMaxVideoSendBandwidth()

Get 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).

getMaxVideoSendBandwidth

void getMaxVideoSendBandwidth(Continuation<Integer> cont)

Get 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).

getMaxVideoSendBandwidth

TFuture<Integer> getMaxVideoSendBandwidth(Transaction tx)

Get 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).

setMaxVideoSendBandwidth

void setMaxVideoSendBandwidth(int maxVideoSendBandwidth)

Set 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).

setMaxVideoSendBandwidth

void setMaxVideoSendBandwidth(int maxVideoSendBandwidth,
                              Continuation<Void> cont)

Set 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).

setMaxVideoSendBandwidth

void setMaxVideoSendBandwidth(int maxVideoSendBandwidth,
                              Transaction tx)

Set 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).

getMediaState

MediaState getMediaState()

Get Media flow state.

• CONNECTED: There is an RTCP flow.
• DISCONNECTED: No RTCP packets have been received for at least 5 sec.

getMediaState

void getMediaState(Continuation<MediaState> cont)

Get Media flow state.

• CONNECTED: There is an RTCP flow.
• DISCONNECTED: No RTCP packets have been received for at least 5 sec.

getMediaState

TFuture<MediaState> getMediaState(Transaction tx)

Get Media flow state.

• CONNECTED: There is an RTCP flow.
• DISCONNECTED: No RTCP packets have been received for at least 5 sec.

getConnectionState

ConnectionState getConnectionState()

Get Connection state.

• CONNECTED
• DISCONNECTED

getConnectionState

void getConnectionState(Continuation<ConnectionState> cont)

Get Connection state.

• CONNECTED
• DISCONNECTED

getConnectionState

TFuture<ConnectionState> getConnectionState(Transaction tx)

Get Connection state.

• CONNECTED
• DISCONNECTED

getMtu

int getMtu()

Get 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.

getMtu

void getMtu(Continuation<Integer> cont)

Get 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.

getMtu

TFuture<Integer> getMtu(Transaction tx)

Get 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.

setMtu

void setMtu(int mtu)

Set 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.

setMtu

void setMtu(int mtu,
            Continuation<Void> cont)

Set 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.

setMtu

void setMtu(int mtu,
            Transaction tx)

Set 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.

getRembParams

RembParams getRembParams()

Get Advanced parameters to configure the congestion control algorithm.

getRembParams

void getRembParams(Continuation<RembParams> cont)

Get Advanced parameters to configure the congestion control algorithm.

getRembParams

TFuture<RembParams> getRembParams(Transaction tx)

Get Advanced parameters to configure the congestion control algorithm.

setRembParams

void setRembParams(RembParams rembParams)

Set Advanced parameters to configure the congestion control algorithm.

setRembParams

void setRembParams(RembParams rembParams,
                   Continuation<Void> cont)

Set Advanced parameters to configure the congestion control algorithm.

setRembParams

void setRembParams(RembParams rembParams,
                   Transaction tx)

Set Advanced parameters to configure the congestion control algorithm.

addMediaStateChangedListener

ListenerSubscription addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener)

Add a EventListener for event MediaStateChangedEvent. Synchronous call.

Parameters:

listener - Listener to be called on MediaStateChangedEvent

Returns:

ListenerSubscription for the given Listener

addMediaStateChangedListener

void addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener,
                                  Continuation<ListenerSubscription> cont)

Add a EventListener for event MediaStateChangedEvent. Asynchronous call. Calls Continuation<ListenerSubscription> when it has been added.

Parameters:

listener - Listener to be called on MediaStateChangedEvent

cont - Continuation to be called when the listener is registered

removeMediaStateChangedListener

void removeMediaStateChangedListener(ListenerSubscription listenerSubscription)

Remove a ListenerSubscription for event MediaStateChangedEvent. Synchronous call.

Parameters:

listenerSubscription - Listener subscription to be removed

removeMediaStateChangedListener

void removeMediaStateChangedListener(ListenerSubscription listenerSubscription,
                                     Continuation<Void> cont)

Remove a ListenerSubscription for event MediaStateChangedEvent. Asynchronous call. Calls Continuation<Void> when it has been removed.

Parameters:

listenerSubscription - Listener subscription to be removed

cont - Continuation to be called when the listener is removed

addConnectionStateChangedListener

ListenerSubscription addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener)

Add a EventListener for event ConnectionStateChangedEvent. Synchronous call.

Parameters:

listener - Listener to be called on ConnectionStateChangedEvent

Returns:

ListenerSubscription for the given Listener

addConnectionStateChangedListener

void addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener,
                                       Continuation<ListenerSubscription> cont)

Add a EventListener for event ConnectionStateChangedEvent. Asynchronous call. Calls Continuation<ListenerSubscription> when it has been added.

Parameters:

listener - Listener to be called on ConnectionStateChangedEvent

cont - Continuation to be called when the listener is registered

removeConnectionStateChangedListener

void removeConnectionStateChangedListener(ListenerSubscription listenerSubscription)

Remove a ListenerSubscription for event ConnectionStateChangedEvent. Synchronous call.

Parameters:

listenerSubscription - Listener subscription to be removed

removeConnectionStateChangedListener

void removeConnectionStateChangedListener(ListenerSubscription listenerSubscription,
                                          Continuation<Void> cont)

Remove a ListenerSubscription for event ConnectionStateChangedEvent. Asynchronous call. Calls Continuation<Void> when it has been removed.

Parameters:

listenerSubscription - Listener subscription to be removed

cont - Continuation to be called when the listener is removed