make and manage custom applications

[Evgeni Baldzhiyski]

Hi Flashphoner team,

First I want to say I am really impressed from the tool you made and really enjoy to use it.

Currently I am trying to prepare solution for our company before they are going to buy license but I stuck in one point and I need your help:

I need to make our own application redirected to our own REST API server and in the java part have to be copy of the currently existed application "flashStreamingApp". I saw CLI allows to be updated/added main and handler app classes but I haven't been able to see the main class mane on the application "flashStreamingApp". Maybe I miss something but I was not able to see any guide about this. Can you please give me the class names of the build in applications. Thanks.

And one more question I have:
When I make sip to rtmp call I saw the rtmp audio stream is coded
Stream #0:0: Audio: aac (LC), 44100 Hz, mono, fltp
I am talking about when I use "rtmpUrl":
{
"appKey":"defaultApp",

"callId": "123456789",
"callee":"test",
"hasAudio": "true",
"hasVideo": "false",

"rtmpUrl": "rtmp://**********:1935/test/instance_name",
"rtmpStream": "stream23",

"sipAuthenticationName":"***",
"sipDomain":"****",
"sipOutboundProxy": "***.",
"sipPort":"5060",
"sipLogin":"test",
"sipPassword":"1234567890aA",

"sipRegisterRequired":"false"
}

Our media conference room server is Adobe Media Server and when I try to play the stream provided from WCS it is very choppy and bad quality.
I tried something other:
{
"appKey":"defaultApp",

"callId": "123456789",
"callee":"test",
"hasAudio": "true",
"hasVideo": "false",
"toStream": "*flv:stream23",

"sipAuthenticationName":"***",
"sipDomain":"***",
"sipOutboundProxy": "***",
"sipPort":"5060",
"sipLogin":"test",
"sipPassword":"1234567890aA",

"sipRegisterRequired":"false"
}
And then I made an our AMS to play the rtmp stream from WCS. The result stream was coded
Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
I played it again with the flash player client and it was very nice sounded.

My question is do I have any option to control the codec in the trmp stream when I use "rtmpUrl"?

I hope you I will find time to answer me fast.
Thank you very much

 

[Evgeni Baldzhiyski]

I found the app classes. Sorry for the stupid question.
But still sty up the question for the codecs.

Thanks

 

[Evgeni Baldzhiyski]

Wow ))) I found solution for the second question too! When is red a stream like this (WCS publish on AMS) is required to set any bufferTime. I hope this will help somebody falls in a case like this.

But still I am curious if there has any way for control the stream codecs.

Thanks

 

[Max]

Good morning.
In your case (SIP as RTMP) you can control RTMP stream codecs via SDP. You should create file WCS_HOME/conf/media_transponder.sdp (its default settings are hardcoded, so file does not exists by default) and set audio codecs in that file.

 

[Evgeni Baldzhiyski]

Hello,

I want to pay attention on the AWS AMI. In the begin I started with FlashPhoner I start with the AMI but first there has been old version 5.0.xxx and second when you try to update it with service webcallserver update it update to version 5.0.34 (or 33 I don't remember very well) and say this is up to day version. In the end after a lot strange problems I was forced to give up and to download the manual installer.
My suggestion either you update the AMI to the latest FlashPhoner version or just remove this option from your download page. This will safe the tame for next that try to use FlashPhoner.

And again I want to give you my congratulations for the awesome product you have made.
Thanks

 

[Max]

Thanks for the kind words, we try to make our product useful.
We work on update AWS distribution to most actual version. We let users know when we update it.