In the table below you can see key Web Call Server releases, it’s names, list of new features and bugfixes.






31 July 2017


Stream monitor

Push Call API

MSE fixes


Added Push call API

Added stream monitoring in dashboard

Added sip-as-rtmp-2 demo page

Added New script

Added MP4 VOD test

Added Set embedded player as default demo sample page

Added Embed Player in Stream stat

Added Audio and video time chart and stream uptime

Added View push streams in statistics

Added sip-as-rtmp-3 demo based on REST 2.0 API


Fixed Usage of SipPortManager for SIP ports management

Fixed AudioProcessor record pcm data

Fixed AAC crash with low sample rate and high bitrate

Fixed AAC streaming proxy

Fixed StreamingEngine TS drop for webrtc

Fixed Crazy streamer

Fixed SIP call failed by ICE keepalive

Fixed Enable CORS for new REST API

Fixed MediaTransponder G711 issues

Fixed Flash AAC doubling packets

Fixed onVideoFormat breaks rooms

Fixed Flash, send AAC config twice

Fixed AACStreamEncoder reset synchronization on ts drop

Fixed SIP call hangup reason

Fixed Use one thread for send statistic on manager

Fixed Strip authentication from /client2/** in startWithDemoUser mode

Fixed MSE sdp creation, use hasAudio/hasVideo for Flash only

Fixed Buffer overflow on inject sound.

Fixed NullPointer on stop playing stream

Fixed No audio in flash to flash streaming

Fixed SessionKeepAliveThread – check full playing or publishing status including intermediary statuses NOT_ENOUGH_BANDWIDTH, RESIZE, etc

Fixed sip-as-rtmp a\v synch

Fixed Play recorded files on ios devices (add ranges of bytes on http requests)

Fixed Profiling statistics and logs in packet encoder

Fixed Remove stats if stream stopped

Fixed More informative logs on closed WS client.

Fixed No audio with trial license

Fixed Fix record data to files ans small optimization in video packet manipulation

Fixed AAC ffmpeg freeze with 22050 sample rate


Set by default use_new_aac_encoder=true

Set by default disable_streaming_proxy_aac=false



27 June 2017


Embedded player and bugfixes


Added Custom avcc relay to pass data to MSE provider

Added embed player in admin panel

Added AACStreamEncoder


Fixed RTCP synchronization enabling

Fixed OnDataEvent for native flash

Fixed Incoming DataPacket buffer

Fixed construct room state on join

Fixed Set names for timers.

Fixed Room, no video for new participant

Fixed Send status PLAYING after session init

Fixded Prevent concurrent room creation with identical names

Fixed FFOutputWriter ensure audio sync is monotonic

Fixed AudioProcessor ensure audio sync is monotonic

Fixed Calculate AAC encoding sync time based on frames rather than PTIME

Fixed RTSP over HLS stream naming

Fixed Limit video encoder bitrate to 100 Mbps.

Fixed Android hls playback

Fixed Special symbols in rtsp stream uri over HLS

Fixed Connection status logging, stream status logging.

Fixed Handle ICE timeout event.

Fixed Handle DTLS failed event.

Fixed Stop stream by ICE keepalives.

Fixed AAC synch time

Fixed RTSP, don’t send GET_PARAMETER if it not allowed in 200 OK on OPTIONS request

Fixed RTMP session disconnected by keepalive if rtmp excluded from keep_alive.enabled property

Fixed synchronization for webrtc as rtmp with aac codec

Fixed RTMP connection, parse appKey from query string if it was not set in NetConnection.connect()

Fixed CustomAvccRelay use TimestampShifter

Fixed CustomAvccRelay record stream to file

Fixed Added sessionId to SDR.log as latest field

Fixed Set audio port 0 in local sdp for MSE provider if there is no audio in feeding media session (proxy)

Fixed If enable_extended_logging=false, do not save client logs .report

Fixed Resolve X-Frame-Options for embed player

Fixed FFDecoder/Encoder rework jclass cache

Fixed Reset SPS/PPS marker bit.

Fixdd Send rtcp receiver report packet in RTSP interleaved mode.

Fixed Add change log level over rest and in connect for client


Added setting rtsp_user_agent

Added setting ice_authorize_by_address

Added setting value generate_av_for_ua=all

Added setting record_rotation

Added Set sip_as_rtmp_java_client, use_rtmp_java_client to true by default



22 May 2017



 New H.264 processing


Added MP4 reader for debug H.264+AAC streams

Added Call injectSound method

Added Video resolution to MediaTransponder API

Added HLS as RTSP

Added New h264 stream processing

Added HTTPS support for HLS


Fixed FU-A depacketization with broken first bit

Fixed RtmpPublisher connection to rtmp server in separate thread

Fixed RtmpPublisher connect failed notification

Fixed WriteException on rtmp stopped stream

Fixed FFAudioGenericCodec synchronize initialization

Fixed Added unique constraint for App.appKey field. Updating or Removing apps using appKey.

Fixed stream_record_policy=template with constraints:{audio:true, video:false}. NPE.

Fixed push api issues

Fixed NetStream.Unpublish.Success for rtmp publishing over ffmpeg

Fixed ‘service webcallserver update’ command

Fixed bin/ config is applied for ./ and ./ scripts and for WCS core

Fixed Rewrite “custom” object on playStream/publishStream

Fixed Flash handlers do not send status to REST

Fixed Unsupported rtmp command getStreamLength

Fixed Disconnect flash if REST failed.

Fixed Clean subscribe stream map on terminating subscribed stream.

Fixed Delete hls folder on start server

Fixed Add CORS headers for HLS

Fixed RTMFP always start video transmission with an I-Frame

Fixed Remove auth headers from REGISTER when doing unregister.

Fixed Clone packet when transcoding

Fixed Lower h264_max_nalu_size


Added setting aac_bitrate

Added setting use_rtmp_java_client

Added manager setting -Drest_template.user_agent

Added SDP to clientExclude and restExclude by default

Added Moving java properties to conf directory.

Added setting suppress_audio