public interface BaseRtpEndpoint extends SdpEndpoint
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
#mediaState
and the #connectionState
properties. It is
also possible subscribe to events fired when these properties change.
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
#MediaFlowInStateChange
and
#MediaFlowOutStateChange
, 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.
Possible values are:
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 is done here:
b={modifier}:{value}
tag). Kurento will always enforce bitrate
limitations specified by the remote peer over internal configurations.
All bandwidth control parameters must be changed before the SDP negotiation takes place, and can't be changed afterwards.
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 Maximum video bitrate sent to remote peer.
|
void |
getMaxVideoSendBandwidth(Continuation<Integer> cont)
Get Maximum video bitrate sent to remote peer.
|
TFuture<Integer> |
getMaxVideoSendBandwidth(Transaction tx)
Get Maximum video bitrate sent to remote peer.
|
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 bitrate requested on the received video stream.
|
void |
getMinVideoRecvBandwidth(Continuation<Integer> cont)
Get Minimum bitrate requested on the received video stream.
|
TFuture<Integer> |
getMinVideoRecvBandwidth(Transaction tx)
Get Minimum bitrate requested on the received video stream.
|
int |
getMinVideoSendBandwidth()
Get Minimum video bitrate sent to remote peer.
|
void |
getMinVideoSendBandwidth(Continuation<Integer> cont)
Get Minimum video bitrate sent to remote peer.
|
TFuture<Integer> |
getMinVideoSendBandwidth(Transaction tx)
Get Minimum video bitrate sent to remote peer.
|
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 Maximum video bitrate sent to remote peer.
|
void |
setMaxVideoSendBandwidth(int maxVideoSendBandwidth,
Continuation<Void> cont)
Set Maximum video bitrate sent to remote peer.
|
void |
setMaxVideoSendBandwidth(int maxVideoSendBandwidth,
Transaction tx)
Set Maximum video bitrate sent to remote peer.
|
void |
setMinVideoRecvBandwidth(int minVideoRecvBandwidth)
Set Minimum bitrate requested on the received video stream.
|
void |
setMinVideoRecvBandwidth(int minVideoRecvBandwidth,
Continuation<Void> cont)
Set Minimum bitrate requested on the received video stream.
|
void |
setMinVideoRecvBandwidth(int minVideoRecvBandwidth,
Transaction tx)
Set Minimum bitrate requested on the received video stream.
|
void |
setMinVideoSendBandwidth(int minVideoSendBandwidth)
Set Minimum video bitrate sent to remote peer.
|
void |
setMinVideoSendBandwidth(int minVideoSendBandwidth,
Continuation<Void> cont)
Set Minimum video bitrate sent to remote peer.
|
void |
setMinVideoSendBandwidth(int minVideoSendBandwidth,
Transaction tx)
Set Minimum video bitrate sent to remote peer.
|
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.
|
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
addMediaSessionStartedListener, addMediaSessionStartedListener, addMediaSessionTerminatedListener, addMediaSessionTerminatedListener, removeMediaSessionStartedListener, removeMediaSessionStartedListener, removeMediaSessionTerminatedListener, removeMediaSessionTerminatedListener
addElementConnectedListener, addElementConnectedListener, addElementDisconnectedListener, addElementDisconnectedListener, addMediaFlowInStateChangeListener, addMediaFlowInStateChangeListener, addMediaFlowOutStateChangeListener, addMediaFlowOutStateChangeListener, 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, removeMediaFlowInStateChangeListener, removeMediaFlowInStateChangeListener, removeMediaFlowOutStateChangeListener, removeMediaFlowOutStateChangeListener, removeMediaTranscodingStateChangeListener, removeMediaTranscodingStateChangeListener, setAudioFormat, setAudioFormat, setAudioFormat, setMaxOuputBitrate, setMaxOuputBitrate, setMaxOuputBitrate, setMaxOutputBitrate, setMaxOutputBitrate, setMaxOutputBitrate, setMinOuputBitrate, setMinOuputBitrate, setMinOuputBitrate, setMinOutputBitrate, setMinOutputBitrate, setMinOutputBitrate, setOutputBitrate, setOutputBitrate, setOutputBitrate, setVideoFormat, setVideoFormat, setVideoFormat
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
isCommited, release, release, release, waitCommited, whenCommited, whenCommited
int getMinVideoRecvBandwidth()
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.
void getMinVideoRecvBandwidth(Continuation<Integer> cont)
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.
TFuture<Integer> getMinVideoRecvBandwidth(Transaction tx)
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.
void setMinVideoRecvBandwidth(int minVideoRecvBandwidth)
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.
void setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Continuation<Void> cont)
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.
void setMinVideoRecvBandwidth(int minVideoRecvBandwidth, Transaction tx)
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.
int getMinVideoSendBandwidth()
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.
void getMinVideoSendBandwidth(Continuation<Integer> cont)
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.
TFuture<Integer> getMinVideoSendBandwidth(Transaction tx)
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.
void setMinVideoSendBandwidth(int minVideoSendBandwidth)
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.
void setMinVideoSendBandwidth(int minVideoSendBandwidth, Continuation<Void> cont)
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.
void setMinVideoSendBandwidth(int minVideoSendBandwidth, Transaction tx)
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.
int getMaxVideoSendBandwidth()
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
void getMaxVideoSendBandwidth(Continuation<Integer> cont)
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
TFuture<Integer> getMaxVideoSendBandwidth(Transaction tx)
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
void setMaxVideoSendBandwidth(int maxVideoSendBandwidth)
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
void setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Continuation<Void> cont)
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
void setMaxVideoSendBandwidth(int maxVideoSendBandwidth, Transaction tx)
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 parameter to higher values such as 2000 (2 mbps) or even 10000 (10 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
.
MediaState getMediaState()
void getMediaState(Continuation<MediaState> cont)
TFuture<MediaState> getMediaState(Transaction tx)
ConnectionState getConnectionState()
void getConnectionState(Continuation<ConnectionState> cont)
TFuture<ConnectionState> getConnectionState(Transaction tx)
int getMtu()
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.
void getMtu(Continuation<Integer> cont)
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.
TFuture<Integer> getMtu(Transaction tx)
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.
void setMtu(int mtu)
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.
void setMtu(int mtu, Continuation<Void> cont)
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.
void setMtu(int mtu, Transaction tx)
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.
RembParams getRembParams()
void getRembParams(Continuation<RembParams> cont)
TFuture<RembParams> getRembParams(Transaction tx)
void setRembParams(RembParams rembParams)
void setRembParams(RembParams rembParams, Continuation<Void> cont)
void setRembParams(RembParams rembParams, Transaction tx)
ListenerSubscription addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener)
EventListener
for event MediaStateChangedEvent
. Synchronous call.listener
- Listener to be called on MediaStateChangedEventvoid addMediaStateChangedListener(EventListener<MediaStateChangedEvent> listener, Continuation<ListenerSubscription> cont)
EventListener
for event MediaStateChangedEvent
. Asynchronous call.
Calls Continuation<ListenerSubscription> when it has been added.listener
- Listener to be called on MediaStateChangedEventcont
- Continuation to be called when the listener is registeredvoid removeMediaStateChangedListener(ListenerSubscription listenerSubscription)
ListenerSubscription
for event MediaStateChangedEvent
. Synchronous call.listenerSubscription
- Listener subscription to be removedvoid removeMediaStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont)
ListenerSubscription
for event MediaStateChangedEvent
. Asynchronous call.
Calls Continuation<Void> when it has been removed.listenerSubscription
- Listener subscription to be removedcont
- Continuation to be called when the listener is removedListenerSubscription addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener)
EventListener
for event ConnectionStateChangedEvent
. Synchronous call.listener
- Listener to be called on ConnectionStateChangedEventvoid addConnectionStateChangedListener(EventListener<ConnectionStateChangedEvent> listener, Continuation<ListenerSubscription> cont)
EventListener
for event ConnectionStateChangedEvent
. Asynchronous call.
Calls Continuation<ListenerSubscription> when it has been added.listener
- Listener to be called on ConnectionStateChangedEventcont
- Continuation to be called when the listener is registeredvoid removeConnectionStateChangedListener(ListenerSubscription listenerSubscription)
ListenerSubscription
for event ConnectionStateChangedEvent
. Synchronous call.listenerSubscription
- Listener subscription to be removedvoid removeConnectionStateChangedListener(ListenerSubscription listenerSubscription, Continuation<Void> cont)
ListenerSubscription
for event ConnectionStateChangedEvent
. Asynchronous call.
Calls Continuation<Void> when it has been removed.listenerSubscription
- Listener subscription to be removedcont
- Continuation to be called when the listener is removedCopyright © 2021 Kurento. All rights reserved.