The properties are stored in the system registry:
HKEY_CURRENT_USER\SOFTWARE\Medialooks\WebRTC
| Name | Default value | Description |
|---|---|---|
| external_process | true |
Enable external process for WebRTC object. Each WebRTC object will create a new MServer process. This could be useful in terms of reliability - if something goes wrong with you WebRTC object this will not affect your primary application
|
| decoder.gpu_enabled | false | Enable an ability to use GPU for decoding |
| object_name | MWebRTC | |
| signaling_server | https://rtc.medialooks.com:8889 |
By default, we are using our public signaling server, please do not try to use https://rtc.medialooks.com:8889 in the production environment, it was provided
only for testing purpose. For production, you should use your own signaling server
|
| turn_server |
TURN server, should looks like "turn:<your_turn_ip>:<your_turn_port>"
|
|
| turn_user |
User for TURN server
|
|
| turn_password |
Password for TURN server
|
|
| stun_server | stun:stun.l.google.com:19302 | |
| force_native_format | true |
Save native resolution for input frames.
The current property will force WebRTC adaptive bitrate feature to save the original frame resolution.
Please note current property do not disable adaptive bitrate feature, only frame resolution changes.
With "force_native_format"="false", video resolution will be change according to current network bandwith.
|
| enable_audio | true |
Sending Audio
|
| enable_video | true |
Sending Video
|
| login_timeout | dword:00000fa0 |
Signaling server timeout in ms
|
| video_timeout | dword:00000000 |
Timeout video frame on input in ms
If you are going to use audio only streams, you should set current value to 0.
|
| room | Room670 |
Random generated after the first WebRTC running after SDK installation
|
| name | LocalClient8219 |
Peer name by default (random generation after the first WebRTC running after SDK installation)
|
| mode | Any |
Possible Values:
|
| video_encoder | gpu_h264 |
Possible Values:
|
| gpu_h264_interlace | Progressive | By defaul all interlaced video converts to progressive one. |
| audio_encoder | opus |
Possible Values:
|
| audio_processing | false |
Enable/disable built in Google Audio filters.
At the current moment, false value will provide a good sound quality, but will disable some additional features such as "echo cancellation"
Check out current documentation page for updates relative to audio_processing property.
|
| audio_processing.EchoCancellation | false | |
| audio_processing.AutoGainControl | false | |
| audio_processing.NoiseSuppression | false | |
| audio_processing.HighpassFilter | false | |
| audio_processing.StereoSwapping | false | |
| audio_processing.AudioJitterBufferMaxPackets | dword:00000000 | |
| audio_processing.AudioJitterBufferFastAccelerate | false | |
| audio_processing.AudioJitterBufferMinDelayMS | dword:00000000 | |
| audio_processing.TypingDetection | false | |
| audio_processing.ExperimentalAGC | false | |
| audio_processing.ExtendedFilterAEC | false | |
| audio_processing.DelayAgnosticAEC | false | |
| audio_processing.ExperimentalNS | false | |
| audio_processing.ResidualEchoDetector | false | |
| audio_processing.TXAGCTargetDBOV | dword:00000000 | |
| audio_processing.TXAGCDigitalCompressionGain | dword:00000000 | |
| audio_processing.TXAGCLimiter | false | |
| audio_processing.CombinedAudioVideoBWE | false | |
| audio_processing.AudioNetworkAdaptor | false | |
| audio_processing.AudioNetworkAdaptorConfig | ||
| audio_multichannel | true | |
| video_bitrate_min | 300K |
Minimal video bitrate, in case of gpu_h264 peers wich are not able to provide desirable bitrate due network bandwidth limitation will be disconnected
|
| video_bitrate | 5M |
Video Bitrate
|
| audio_bitrate | 128K |
Audio Bitrate
|
| signaling.old_versions_support | true | Use only if you are using an old version of signaling server |
| signaling.reconnect_timeout_msec | dword:00000bb8 | |
| reconnect_attempts | dword:00000014 |
Reconnect attempts count to peer
|
| reconnect_interval | dword:000003e8 |
Reconnect attempts interval in miliseconds
|
| embed_timecode | false |
Frames timecode overlay
|
| embed_closedcaptions | false |
Adding subtitles to frames
|
| statictics_extra |
If "statictics_nofilter"="false" but you would like to add any extra nodes which are removed by default stat filter, you can add it here divided by space
|
|
| statictics_interval | dword:000004b0 |
Interval for getting stats information from WebRTC object in ms
|
| statictics_nofilter | false |
Removes all filters from statistic information
|
| optimize_fps | false |
Performance optimization in case of sending similar frames. This could be useful for example when you are using the ScreenCapture as a source for WebRTC sender.
i.e when you are using ScreenCapture with the following property capture.dxgi_optimize_copy = true (by default) and WebRTC "optimize_fps"="true", while there is no any movements or changes on captured screen, WebRTC will send the same last copied frame 1 per second.
|
| receiver_fps | auto | |
| multicast_type |
Multicast server type
|
|
| multicast_server |
Setting multicast server (wowza/janus) IP address
|
|
| ice_transports | all |
Ice_transports
|
| ice_ipv6_enabled | true | Enable IPv6 for ice transport |
| sdp.transport_cc_enabled | false | Use an old SDP format for web clients, set true only if you are using really old browsers versions. Pay attention it's not recommended to use set the property true by default, only for case old browsers receivers. |
| buffer.min_msec | dword:00000032 |
WebRTC packer minimum value
|
| buffer.max_msec | dword:000000c8 |
WebRTC packer maximum value
Tuning packer buffer size could be useful in situation when you publisher/ receiver not able to provide desirable FPS value due the network issues. Pay attention increasing the buffer value will increase the delay.
|
| constant_quality.min_fps | 0.0 |
Useful in case of network/performance issues, the encoder (publisher side) will try to decrease the quality until desirable minimum FPS value will not be achieved on the receiver side. |
| decoder.nvidia | true | Use NVIDIA GPU for decoding |
| decoder.quicksync | true | Use Intel GPU for decoding |
| cpu_adaptation | false | Enable WebRTC CPU Overuse Detection. This flag comes from the PeerConnection constraint 'googCpuOveruseDetection'. |
| suspend_below_min_bitrate | false | True if the stream should be suspended when the available bitrate fall below the minimum configured bitrate. If this variable is false, the stream may send at a rate higher than the estimated available bitrate. Enabling suspend_below_min_bitrate will also enable pacing and padding, otherwise, the video will be unable to recover from suspension. |
| enable_dscp | false | Setting the googDSCP flag. |