DTMF not working in Sip as Rtmp example

Max

Administrator
Staff member
Could you please zip and send
Code:
/usr/local/FlashphonerWebCallServer/conf
/usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log
/usr/local/FlashphonerWebCallServer/logs/flashphoner_manager.log
to logs@flashphoner.com
We will check.
 

Niks

New Member
Hi Max,
DTMF signal is working in Phone UI example, do I need to implement a voice menu at SIP side for working DTMF in SIP As Rtmp example?
 

Max

Administrator
Staff member
do I need to implement a voice menu at SIP side for working DTMF in SIP As Rtmp example?
How do you test DTMF?

When you call, for example to airport, the automatic voice says 'Press 1 to ...'.
It is IVR (voice menu) and it of course should work on SIP side.

For SIP as RTMP case, DTMF works over REST API

1. You can setup a call:
Code:
https://host:8888/rest-api/call/startup
{
...
}
2. You can find this call by callId and check if the call is established
https://host:8888/rest-api/call/find
Code:
{
...
}
3. Once the call has been established, you can send DTMF
https://host:8888/rest-api/call/send_dtmf
Code:
{
...
}
Please checkout our REST API examples:
https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-rest-api/index.html?examples.htm

You can test these examples using Advanced REST Console
Here you can see the screenshot:
https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-rest-api/index.html?api_urls.htm

Please provide logs listed above.
We need also traffic dump
Code:
tcpdump udp -s 4096 -w log.pcap
Eventually we need the following logs and dump to check this:
Code:
/usr/local/FlashphonerWebCallServer/conf
/usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log
/usr/local/FlashphonerWebCallServer/logs/flashphoner_manager.log
log.pcap
 

Niks

New Member
Hi Max,

I am using your Web Call Server - Developer Guide SIP As RTMP Example, attaching screenshots from where I took reference and from GitHub I got (Sip as rtmp.js) and (sip as rtmp.html)




 

Attachments

Max

Administrator
Staff member
We have two REST API versions, 1.0 and 2.0
This example using version 1.0 and it is deprecated.

Please use REST API 2.0.
You can find docs here:
https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-rest-api/index.html
You can test this API using Advanced REST Console
https://chrome.google.com/webstore/detail/advanced-rest-client/hgmloofddffdnphfgcellkdfbfbjeloo

We plan to update developer guide with REST API 2.0 example.
But currently this example is out of date.

You can also use example SIP as RTMP 2, based on new REST API 2.0
https://wcs5-eu.flashphoner.com/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.html
 

Max

Administrator
Staff member
Test description
1. We use SIP as RTMP 2 example
https://wcs5-eu.flashphoner.com/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.html
2. We use Bria 4 SIP softphone and OpenSIPs as a signaling server.
Therefore we call from WCS to Bria and then forward stream to RTMP server.

Test steps
1. Fill out SIP details and press 'Call'.
2. Once the call is ESTABLISHED, fill out RTMP details and press 'Start'.
3. Once 'Start' label changed to 'Stop', press 'Play' on built-in RTMP player to play stream.

Amazon EC2 settings for SIP as RTMP case
1. Make sure you have latest available build 2358.
Code:
service webcallserver update
2. If you are using Amazon EC2, add following setting
Code:
client_mode=false
in WCS_HOME/conf/flashphoner.properties
3. Make sure you have this line in flashphoner.properties
Code:
rtmp_transponder_stream_name_prefix =rtmp_
Notes
Note that stream name has prefix rtmp_
So if stream is set to 'stream1', it will be re-published as 'rtmp_stream1'
 

Niks

New Member
Hi Max,

I tried once again it is giving the same error, I am doing it for only audio.
I am attaching snapshots. plz, check if I am doing something wrong.




 

Attachments

Max

Administrator
Staff member
Could you please provide full log output.
Or full screenshot with errors.
 
Top