Stream is not playing

[awv]

I rolled back to Asterisk 17, then 16.8, and stream is still silent. I have been running Asterisk 16 or 17 for +2 years and only tried v. 18 lately to solve issue.
I was going to ask if it is doable to roll back WCS easily? But you have replaced WCS in the flow with softphone and PBX still sends silence.

 

[Max]

Hello

We have seen DTMF error

WCS:

08:22:38,400 INFO         RestApiRouter - HTTP-pool-3-thread-193 Use controller class com.flashphoner.rest.server.rest_v2.RestCallController with path /rest-api/call/send_dtmf
08:22:38,400 INFO    RestCallController - HTTP-pool-3-thread-193 handleRequest /rest-api/call/send_dtmf params:{callId=5620-7001, dtmf=1, type=RFC2833}

Asterisk:

[2021-09-29 04:22:53] VERBOSE[4806][C-0000042c] app_read.c: User entered nothing.

It seems DTMF broken on WCS end or not recognized on Asterisk.

As an option you can try to switch DTMF to INFO mode on WCS

1. Switch WCS to DTMF INFO

flashphoner.properties

add line

dtmf=INFO

2. Restart WCS

If Asterisk supports DTMF INFO mode, it should work. We will also check DTMF sending from our end. If it is broken, we will raise a ticket.

Regarding WCS rollback. Yes you can do that easily.

1. Backup WCS folder (copy content from) /usr/local/FlashphonerWebCallServer
2. Stop WCS server
service webcallserver stop
3. Download previous stable version, unzip and run ./install.sh to downgrade.

 

[awv]

I have changed to DTMF Info on Asterisk but error remains. Not sure what is going on with that.
I rolled back to WCS 5.2.1015. I am now able to get audio inside Safari 15 when I change User-Agent to FireFox. No other browser or user-agent seems to get any audio. This confirms to me that Asterisk is indeed sending audio, not silence, but something must be broken in player code somewhere perhaps?

 

[Max]

I have changed to DTMF Info on Asterisk but error remains. Not sure what is going on with that.

You should change this in add_wcs_to_conference.sh script too:

curl -s -X POST -d "{ \"callId\": \"5620-$CONF_NUMBER\", \"dtmf\": \"1\", \"type\": \"INFO\" }" "$URL/send_dtmf"

We've done it, and the error gone, Asterisk displays in logs

[2021-09-29 23:13:10] VERBOSE[19955][C-00000301] app_read.c: User entered '1'

But we still do not listen anything from softphone which is calling to 7002, may be we shoud also enter a pin?

I rolled back to WCS 5.2.1015. I am now able to get audio inside Safari 15 when I change User-Agent to FireFox. No other browser or user-agent seems to get any audio. This confirms to me that Asterisk is indeed sending audio, not silence, but something must be broken in player code somewhere perhaps?

If you're using player2, you should also fix relative paths in player_audio.html and player.js. We see the errors in any browser console on player page:

This means all the paths like ../../../../flashphoner.js should be changed to absolute paths, for example https://flash.polycall.net:8444/client2/flashphoner.js and so on.

 

[awv]

Thanks for fixing the dtmf issue!
I am using another player, this player, I did update all the links to be sure they were absolute. I do see several errors (Edge browser developer tools lists them nicely) and wondered if this player will recieve updates to be compatible with other modern browsers. Presently we can get audio only on Firefox browsers. Can you duplicate that issue? I'm not sure when this started happening, I'm quite sure this is fairly recent, but maybe you can check which version affected player scripts?
Regarding the PIN on the softphone, I have the PBX set up to play hold music if only 1 user is present in the conference. If dialing 7002, the stream to WCS and the softphone are both active, and neither has hold music. Hang up on the softphone and the stream should play hold music.
Alternately to check through softphone, ensure stream has ended, dial 7001 through softphone and then 1 to enter the conference (muted) and as no other user is present in the conference, you will hear hold music.
Any other clues we can pursue?

 

[Max]

I am using another player, this player, I did update all the links to be sure they were absolute. I do see several errors (Edge browser developer tools lists them nicely) and wondered if this player will recieve updates to be compatible with other modern browsers. Presently we can get audio only on Firefox browsers. Can you duplicate that issue? I'm not sure when this started happening, I'm quite sure this is fairly recent, but maybe you can check which version affected player scripts?

As we see, this page uses player.js default script to play.

    <script type="text/javascript" src="https://flash.polycall.net:8444/client2/examples/demo/streaming/player/player.js"></script>

This script uses default constraints, and make a browser wait for both audio and video packets to be received. But call stream contains only audio track. So you should copy constraints declaration from player_audio2.js[ICODE] to [ICODE]player.js script:

function playStream(session) {
    var streamName = $('#streamName').val();
    var options = {
        name: streamName,
        display: remoteVideo,
        flashShowFullScreenButton: true,
        constraints: {audio: true, video: false},
        transport: "TCP"
    };
    ...
}

In this case, call stream should play in the most of browsers (we've tested this in Chrome and Edge too).

Regarding the PIN on the softphone, I have the PBX set up to play hold music if only 1 user is present in the conference. If dialing 7002, the stream to WCS and the softphone are both active, and neither has hold music. Hang up on the softphone and the stream should play hold music.

Yes, this works. We also added constraints to the script as mentioned above, and stream is playing hold music.

 

[awv]

That is fantastic! Thanks an awful lot, it all appears to be working as it should be.

 

[tharunram]

Hi Team,

How can we continuosly record our ip camera streams(multiple) with flashphoner and save it.
Note: We are already using flashphoner for HTML5-RTSP player for IP cams ,So we need to store that streams.
Is there any possibility ,if yes should we have purchase any license for that feature to be available?

Thanks,

 

[Max]

Good day.

How can we continuosly record our ip camera streams(multiple) with flashphoner and save it.
Note: We are already using flashphoner for HTML5-RTSP player for IP cams ,So we need to store that streams.
Is there any possibility ,if yes should we have purchase any license for that feature to be available?

You should not buy any additional license because recording features are already included.
You can record all the REST streams captured by adding the following parameter to flashphoner.properties file:

record_rtsp_streams=true

Please also read the details about stream recording here.

 

[awv]

Hello,
Are you able to check my server for this same issue which popped up again? Seems like player is waiting for video packets to be received, but I am only sending audio packets. So player is silent. I am using this player, should be the default player. I do not need the video player to come up, only the start button to play audio.
Somehow I think my player has become corrupted between upgrades. I am running 5.2.1105.

 

[Max]

Are you able to check my server for this same issue which popped up again? Seems like player is waiting for video packets to be received, but I am only sending audio packets. So player is silent. I am using this player, should be the default player. I do not need the video player to come up, only the start button to play audio.

We've checked the player by link:
1. Publish a stream using Two Way Streaming example
2. Play this stream in player
Seems like player contains the recommended modifications and audio is playing correctly:

 

[tharunram]

Hi Max,

As we are using Flash phoner HTML5-RTSP player for IP cams feature in our application ,Can we continuosly record the videos in background even the flashphoner is not accessed?

Thanks,

 

[Max]

As we are using Flash phoner HTML5-RTSP player for IP cams feature in our application ,Can we continuosly record the videos in background even the flashphoner is not accessed?

First, please create your own topic for futher questions.
You can capture RTSP streams and record them using REST API
Stream capture from the IP camera management by REST API
Stream recording on demand