[asterisk-bugs] [JIRA] (ASTERISK-24735) VP8 Pass-through support for (WebRTC endpoints)
Javier Fernando Riveros Antequera (JIRA)
noreply at issues.asterisk.org
Wed Jan 28 14:29:34 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24735?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Javier Fernando Riveros Antequera updated ASTERISK-24735:
---------------------------------------------------------
Description:
If this is a duplicate: sorry for the noise. I failed to find it on this versions.
Test ) Call between two webrtc peers firefox 34 jssip client, asterisk playback audio before dial.
On Playback(letters/asterisk); works great
On Dial ;dial works for chan_sip or chan_pjsip. i only get this warnings
WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Jan 28 17:37:00] WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110
Results
- Audio ulaw works great.
- Video VP8 not work on ast 13.1 ; in ast 11 with patch VP8 pass.. video work with same versions of clients and configs,.
- Signalling seems to be OK. compare with ast11
- This behavior is the same for chan_sip and chan_pjsip.
- Curious thing if you call from softphones (linphone) that support udp vp8 to web browser , linphone could see video from web browser but web browser couldn't see video from linphone, seems like asterisk is changing something on VP8 streams when webrtc peer is involve.
When you call between WebrRTC endpoins Asterisk 13.1 is sending media (audio, video) to both legs of the call but video part not work , Firefox/chrome Video debug said "Received incomplete frame timestamp" and "Decoder error: -1"
DEBUG ; (15: 1:23:345 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 1) VIDEO CODING: 0 1; 8259; ExtrapolateLocalTime(1357020)=22163368 ms
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Render frame 3159780769 at 1357020. Render delay 22163474
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Received incomplete frame timestamp 1353960 frame size 809 at time 22163413
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Packet received and sent to jitter estimate with: timestamp=1353960 wall_clock=22163413
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Jitter estimate updated with: frameSize=809 frameDelayMS=-5
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Framesize statistics: max=1870.187082 average=1469.215404
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; The estimated slope is: theta=(0.002510, 10.590306)
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Random jitter: mean=-2.833788 variance=2419.359935
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current jitter estimate: 85.612250
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current max RTT: 0
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; g1=0.000000 g2=-384.621773 alarm=0
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; w[0]=89.480924 w[1]=-238058.950884 ts=1357020 tMs=17553
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoding timestamp 1357020
ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoder error: -1
ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Failed to decode frame 1357020, requesting key frame
Thanks
was:
If this is a duplicate: sorry for the noise. I failed to find it on this versions.
Test ) Call between two webrtc peers firefox 34 jssip client, asterisk playback audio before dial.
On Playback(letters/asterisk); works great
On Dial ;dial works for chan_sip or chan_pjsip. i only get this warnings
WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Jan 28 17:37:00] WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110
Results
- Audio ulaw works great.
- Video VP8 not work on ast 13.1 ; in ast 11 with patch VP8 pass.. video work with same versions of clients and configs,.
- Signalling seems to be OK. compare with ast11
- This behavior is the same for chan_sip and chan_pjsip.
- Curious thing if you call from softphones (linphone) that support udp vp8 to web browser , linphone could see video from web browser but web browser couldn't see video from linphone, seems like asterisk is changing something on VP8 streams when webrtc peer is involve.
Asterisk 13.1 is sending media (audio, video) to both legs of the call but video part not work , Firefox/chrome Video debug said "Received incomplete frame timestamp" and "Decoder error: -1"
DEBUG ; (15: 1:23:345 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 1) VIDEO CODING: 0 1; 8259; ExtrapolateLocalTime(1357020)=22163368 ms
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Render frame 3159780769 at 1357020. Render delay 22163474
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Received incomplete frame timestamp 1353960 frame size 809 at time 22163413
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Packet received and sent to jitter estimate with: timestamp=1353960 wall_clock=22163413
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Jitter estimate updated with: frameSize=809 frameDelayMS=-5
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Framesize statistics: max=1870.187082 average=1469.215404
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; The estimated slope is: theta=(0.002510, 10.590306)
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Random jitter: mean=-2.833788 variance=2419.359935
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current jitter estimate: 85.612250
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current max RTT: 0
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; g1=0.000000 g2=-384.621773 alarm=0
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; w[0]=89.480924 w[1]=-238058.950884 ts=1357020 tMs=17553
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoding timestamp 1357020
ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoder error: -1
ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Failed to decode frame 1357020, requesting key frame
Thanks
> VP8 Pass-through support for (WebRTC endpoints)
> -----------------------------------------------
>
> Key: ASTERISK-24735
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24735
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip, Channels/chan_sip/SRTP, Channels/chan_sip/Video, Resources/res_srtp
> Affects Versions: 12.8.0, 13.1.0
> Environment: OS: ubuntu 14.04
> Asterisk: 13.1.0 current version.
> Client : jssip 0.6.12 online demo ( disable new session timers feature ) /firefox 34 /chrome 39
> channels : Chan_sip , chan_pjsip
> Reporter: Javier Fernando Riveros Antequera
> Attachments: Ast_Debug_WebRTC-VP8LOG, firefox_debug_output.txt, rtp_ast_13_1_vp8_error.txt, rtp_conf.txt, sip_ast_13_1_vp8_error.txt, sip_conf.txt
>
>
> If this is a duplicate: sorry for the noise. I failed to find it on this versions.
> Test ) Call between two webrtc peers firefox 34 jssip client, asterisk playback audio before dial.
> On Playback(letters/asterisk); works great
> On Dial ;dial works for chan_sip or chan_pjsip. i only get this warnings
> WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
> [Jan 28 17:37:00] WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110
> Results
> - Audio ulaw works great.
> - Video VP8 not work on ast 13.1 ; in ast 11 with patch VP8 pass.. video work with same versions of clients and configs,.
> - Signalling seems to be OK. compare with ast11
> - This behavior is the same for chan_sip and chan_pjsip.
> - Curious thing if you call from softphones (linphone) that support udp vp8 to web browser , linphone could see video from web browser but web browser couldn't see video from linphone, seems like asterisk is changing something on VP8 streams when webrtc peer is involve.
> When you call between WebrRTC endpoins Asterisk 13.1 is sending media (audio, video) to both legs of the call but video part not work , Firefox/chrome Video debug said "Received incomplete frame timestamp" and "Decoder error: -1"
> DEBUG ; (15: 1:23:345 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
> DEBUG ; (15: 1:23:346 | 1) VIDEO CODING: 0 1; 8259; ExtrapolateLocalTime(1357020)=22163368 ms
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Render frame 3159780769 at 1357020. Render delay 22163474
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Received incomplete frame timestamp 1353960 frame size 809 at time 22163413
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Packet received and sent to jitter estimate with: timestamp=1353960 wall_clock=22163413
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Jitter estimate updated with: frameSize=809 frameDelayMS=-5
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Framesize statistics: max=1870.187082 average=1469.215404
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; The estimated slope is: theta=(0.002510, 10.590306)
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Random jitter: mean=-2.833788 variance=2419.359935
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current jitter estimate: 85.612250
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Current max RTT: 0
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; g1=0.000000 g2=-384.621773 alarm=0
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; w[0]=89.480924 w[1]=-238058.950884 ts=1357020 tMs=17553
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 1; 8259; Delay: min_playout=0 jitter=96 max_decode=0 render=10
> DEBUG ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoding timestamp 1357020
> ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Decoder error: -1
> ERROR ; (15: 1:23:346 | 0) VIDEO CODING: 0 0; 8259; Failed to decode frame 1357020, requesting key frame
> Thanks
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