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





12 Nov



Load balancer and re-streaming over WebRTC


Performance fixes


RTSP fixes


Sync fixes


Added LoadBalancingController WebRTC republishing

Added Turn off LoadBalancingController republishing by default

Added use_rtmp_server_socket_timeout server property, default false

Added webrtc_agent_use_webrtc=true property to switch WebRTC push and pull to AVP profile

Added Allow to force rtp synchronization

Added Allow to record sip-as-rtmp call with “sip_as_rtmp_record_stream” property

Added Inject custom http headers to REST

Added settings hls_player_width=640 hls_player_height=480, config

Added settings sip.pre_init=true

Added rest api for snapshots

Added Add rtp-in buffer for rtsp (rtsp_in_buffer)

Added  sip_use_reentrant_listener property

Added setting ice_keep_alive_enabled=true


Fixed LBConfig

Fixed Synchronization issues with stream republishing to WCS server

Fixed Force closing media session if exception was catched on creating it

Fixed Fix usage force synchronization for streams with rtcp

Fixed fixes process incoming data from android chrome and changes from jitter buffer

Fixed Clone stream object on notifyStreamStatusEvent to prevent original object modification

Fixed Json ignore for playing and publishing

Fixed Video distributor thread leak

Fixed Fix ice problem on edge

Fixed Delete context fields from snapshots models

Fixed Rtsp fix on rtcp port in setup request

Fixed Add record_rtsp_streams property and fix rtp synchronization bug (force_rtp_synchronization)

Fixed rtsp create snapshot

Fixed Fix rtsp playing with only video

Fixed Memory usage optimization

Fixed Fix record rtsp streams with pps in sdp

Fixed added rtmp_transponder_full_url to RtmpShadowTransponderJava

Fixed High Level keep alives does not work for WSPlayer

Fixed Fixes in creating snapshots

Fixed Catch possible exception on closing RTCP channel

Fixed Workaround receive nal unit with same timestamp


25 Sep



WebRTC pulling


Async REST hooks


SIP fixes


WebRTC fixes



Added to pool webrtc stream from another WCS server

Added New property process_remote_sdp_candidates = true, enable sdp candidate processing

Added New property ice_consent_freshness = true, send binding request instead of binding indication for consent freshness

Added Create jstack dump on cemetery event

Added WebRTCAgent shutdown

Added stopPullAgent/getPullAgents

Added ‘token’ field to (room app)

Added SIP failover REGISTER

Added read server socket timeout 30 seconds on RTMP/TCP channel

Added rest_template.read_timeout property (manager), how long rest client will wait for rest response in milliseconds

Added rest_template.connection_timeout property (manager), how long rest client will wait for connection establishment in milliseconds before considering connection failed

Added Move from jdbc to HikariCP

Added Setup sdp media attribute for rtc sessions

Added SIP failover calls

Added Passthrough websocket Origin header to REST

Added Handle pre-flight CORS requests


Fixed send CORS if configured for content type “text/plain”

Fixed Rtmp transponder can’t push stream if rtmp url does not have trailing “/”

Fixed Rest client connection and response timeouts

Fixed Async node -> manager interaction

Fixed No audio for SIP call due to synchronization.

Fixed Catch exception on monitoring service

Fixed Close DNS resource after srv lookup

Fixed Ice PaceMaker thread leak

Fixed Minor loadbalancing fixes

Fixed Stats and rtmp republishing

Fixed Fix view mp4 files from dependencies


30 August







Added Outgoing RTMP authentication

Added First version of save and view stream history

Added Basic vod using urls vod://file.mp4 or vod-life://file.mp4

Added Media providers HLS,RTSP,PUSH,VOD,WSPlayer


Fixed Parse outgoing rtmp credentials

Fixed View realtime stat graphics

Fixed Media port statistic

Fixed Delay 1000 ms, no RTCP in first packets WebRTC, use RapidResynchronizationRequest when in AVPF profile to request SenderReport from far end

Fixed Do not play media received via rtp if there is no synchronization on it

Fixed Synchronization and reordering buffer for incoming rtp packets

Fixed Statistic “dead”-streams

Fixed Terminate SIP call on INVITE timeout.

Fixed Remove old stream stat data

Fixed Load balancer HTTP stats renamed

Fixed Fix Streams tables

Fixed Enclose table field ‘in’ within backticks to prevent mysql creation error

Fixed Enclose table Cert field ‘key’ within backticks to prevent mysql creation error

Fixed Rename table fields to prevent using sql keywords

Fixed Fix encoder odd resolution