В таблице представлены названия ключевых релизов и перечислены новые функции и внесенные исправления, начиная с 3 января 2017 г.

Build

Status

Notes

2199

 

1 May 2017

 

RTMP games and new REST API

New

Added new RTMP re-publishing implementation

Added REST API methods such as /push/startup  for RTMP re-publishing

Fixed

Fixed parameter custom for room API

Fixed WebRTC as RTMFP no video

Fixed WebRTC video-only stream recording for Screen Sharing case

Fixed Chrome 58, RTMFP as WebRTC playback with H.264 pathrough (Set avc_remove_sei to true)

Fixed sync time for WebRTC, RTSP, SIP to RTMP/RTMFP  cases

Fixed WebRTC streaming subscribe session leak

Fixed deadlock on web client disconnect during WebRTC session initialization

Fixed websocket thread lock by WebRTC DTLS thread

Fixed WebRTC ICE binding problem with Asterisk SIP callID having «:»

Fixed server crash when video height or video width is odd value in transcoding

Fixed applying of max publishing WebRTC bitrate, set from client SDK

Set rtc_ice_add_local_component to true by default

Set avc_remove_sei to true by default

2158

 

31 March 2017

 

Bitrate Catcher

New  

Added new congestion control algorithm webrtc_cc2 for published WebRTC streams

Added RTMP in buffer implementation

Added global bandwidth stats for server network interfaces

Added NOT_ENOUGH_BANDWIDTH event for stream playback over WebRTC

Added REST API 2.0

Added injecting of custom object from RTMP URL params

Added Letsencrypt support for Amazon EC2 instances

Added license number as a parameter for activation script

Fixed  

Fixed WebRTC blocking threads on DTLS handshake

Fixed AAC for RTMP  re-publishing cases

Fixed WebRTC re-transmit burn, adding bitrate limits

Fixed RTMFP connection leak by keep alive

Fixed WebRTC issue with Chrome 57 using rtcp-mux by default

Fixed SIP call issue with legacy WCS3 client PhoneJS based on Flash

Fixed incoming SIP call issue with legacy WCS3 client  PhoneJS based on WebRTC

Set webrtc_cc2 to true by default

Added setting stream_record_policy_template

Added setting rtmp_transponder_full_url

Added setting rtmp_in_buffer_enabled

Added setting rest_method_extension

Added server.properties setting keep_alive.enabled

2106

 

1 March 2017

 

New ICE and bugfixes under load 

New

Added session debug tool available in Dashboard / Demo

Added new WebRTC ICE implementation

Added snapshots for WebRTC and Flash streams

Added keep alives for RTMP

Fixed

Fixed Amazon EC2 instance scan by disabling SSLv3 support for Java 1.8.031 and newer

Fixed WebRTC-SIP Click to Call bugs on disconnect

Fixed SSL certificates configuration from the Dashboard / Security menu

Fixed SIP UPDATE request flow incrementing cseq

Fixed SIP duplicated Supported:timer header

Fixed threading issue with concurrent reading of H.264 NALUs affecting WebRTC and Flash streams

Fixed RTMP/RTMFP stream name should be unique

Fixed RTSP by terminating RTSP publishing session if it is already exist

Fixed blocked workers by adding DeadlockAware implementation and workers cemetery

Fixed RTMP connection was failed when RTMP URL is too long

Fixed VP8 WebRTC recording by using actual incoming stream resolution

Set rtc_ice_add_local_component to true for Amazon EC2 only

Set client_mode to false for Amazon EC2 only

Set streaming_video_decoder_fast_start to true by default

Added setting sip_force_session_expires

2060

 

1 Feb 2017

 

RTSP and transponders

New

Added UTF-8 encoding support for RTSP credentials

Added demo WebRTC as RTMP

Aded demo WebRTC streaming behind a firewall

Added new implementation for WebRTC, SIP as RTMP using media transponders

Fixed

Fixed RTSP stream playback by fixing sprop parameters in  re-packetizer

Fixed RTSP stream if got empty response SDP

Fixed RTSP should fail if no common codec were found

Fixed RTSP as RTMP stream not found

Fixed RTSP, WebRTC H.264 depacketizer by fixing sps/pps insertion, fu-a, correct seq numbers, stap_a packetization

Fixed RTSP URI validation

Fixed WebRTC VP8 recording no video in webm file

Fixed AAC encoding timestamps

Fixed video playback stability by increasing rtcp sender interval

Fixed web call server update command

Fixed server crashes related JNI caching

Fixed RTMP NetStatusEvent on publish failed

Fixed ability of rejecting of RTMP streams with REST

Fixed sync issues by adding ability to use RTCP synch in RTMP subscribe streams

Fixed RTMP playback issue of incoming stream published from Adobe AMS server

Fixed Firefox to chrome SIP video call issue, no video after hold / unhold

Fixed SIP as RTMP crash on null data from transcoder

Fixed install.sh script

Set rtcp_sender_interval to 0.10

2020

 

3 Jan 2017

 

Stress tests and optimizations

New

Added AAC codec support for SIP as RTMP case

Added simple report tool script

Added WebRTC streaming stress tests

Added video + screen sharing demo

Added HLS config

Fixed

Fixed HLS audio video sync

Fixed H.264 de-packetization by sending PLI on buffer flush

Fixed WebRTC playback, by requesting I-frame for a new subscriber

Fixed WebRTC playback should raise failed event if stream is not yet published

Fixed WebRTC PUBLISHING event in time, after session establishment

Fixed WebRTC transcoding audio quality with codec concealment feature for opus decoding

Fixed RTMFP buffer overflow error

Fixed WebRTC Chrome empty payload on startup

Fixed PLI request throttle

Fixed RTMFP memory allocations

Fixed publishing more than one stream via the room API

Fixed preferring IPv4 ports in manager

Fixed HLS audio only stream

Set codecs to opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv by default

Set streaming_video_decoder_fast_start to false by default