[asterisk-bugs] [JIRA] (ASTERISK-24735) Video Media support broken for (WebRTC endpoints)

Mon Feb 16 12:59:35 CST 2015

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24735?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224953#comment-224953 ] 

Matt Jordan commented on ASTERISK-24735:
----------------------------------------

[~goseeped]:

You need to provide more context for your statements.

{quote}
* We are able to make direct calls between Webrtc Endpoints in some conditions (in some clients and web browser doesn't work).
{quote}

Specifically, what worked? What didn't work? What channel driver were you using? What browsers?

{quote}
* We notice after second patch when video is involved on Webrtc endpoints that Playback of audio files doesn't work either.
{quote}

That's going to be an audio stream issue, and unrelated to the video stream.

[~jvanvleet]:

Setting the {{frame_ending}} integer to non-zero is appropriate for video frames, and shouldn't affect non-video streams:

{code}
struct ast_frame_subclass {
	/*! A frame specific code */
	int integer;
	/*! The asterisk media format */
	struct ast_format *format;
	/*! For video formats, an indication that a frame ended */
	unsigned int frame_ending;
};
{code}

So I'd suspect that your solution is correct, or at least very close to being completely correct.

> Video Media support broken 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, 13.1.1, 13.2.0
>         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, frame.c.diff, res_rtp_asterisk.c.diff, 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
> {noformat}
> WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
> WARNING[10405][C-00000001]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110
> {noformat}
> 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 WebRTC 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"
> {noformat}
> 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
> {noformat}
> Update 29 Jan 2015: this happens for h264 codecs to.
> Thanks 

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