Issue with SIP audio calls in SIP as RTMP .

K

Karan Bhansali

Member
Hi,

We have tried another audio conference bridge, we found that it takes only first number from DTMF and participant cannot join the conference. We have dialed from the SIP as RTMP 2 demo page. We have attached the screenshot below.
In below screenshot you can find dtmf pin code is 77519223# but when we listen to voice it takes only 7 and operator says pic code is invalid.
 

Attachments

Max

Max

Administrator
Staff member
Hello.
Please do the following:
1) Try to call from SIP as RTMP not to Plivo conference, but to some automatic call center with IVR. If it works, then go to step 2. If it does not work, go to step 3.
2) Try to call to Plivo conference via softphone (Bria for example). If it works, go to step 3. If it does not work, then seems like there is a problem with RFC2833 DTMF support on Plivo side, and you should go to the Plivo technical support.
3) In WCS_HOME/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.js file try to change if this function (string 354):
Code: 
function sendDTMF(value) {
    var url = field("restUrl") + "/call/send_dtmf";
    var data = {};
    data.callId = callId;
    data.dtmf = value;
    data.type = "RFC2833";
    data = JSON.stringify(data);
    sendREST(url, data);
}
data.type to INFO or INFO_RELAY:
Code: 
...
data.type = "INFO"
...
Then repeat the step 1.
 
K

Karan Bhansali

Member
Hi Max,
We have tried above solution but not working.
When we sending only single digit dtmf i.e(9#) then it works perfect, it joins conference. Also we changed Appkey value from defalutApp to callApp.
We need your help to solve this, as it taking only one digit dtmf, earlier taking multiple digit.
 
Max

Max

Administrator
Staff member
Hello.
We raised internal ticket (WCS-1712) on this issue and let you know when wi do something.
 
Max

Max

Administrator
Staff member
Hello.
If you make a SIP call with REST API, you should set generate_av_for_ua option to generate audio data. As you mentioned in this topic, you have set it to
generate_av_for_ua=Twilio Media Gateway
Click to expand...
but Plivo does not recognize audio data in this case. You should change it to
generate_av_for_ua=all
Click to expand...
and it should work with Plivo.
 
Max

Max

Administrator
Staff member
Hello.
/rest-api/call/startup returns 409 Conflict if there is already a call with the same callId, toStream or rtmpUrl parameters. Please check it.
 
Max

Max

Administrator
Staff member
Hello.
Is the issue reproduced with latest WCS build (5.2.25) with
Code: 
generate_av_for_ua=all
setting?
 
M

Muhammad Rizwan

Member
hello i have similler issues with DTMF i have installed the build 5.1.3695 but still have same work issue with DTMF it is not working. i have tried calling gradwell company number 441225800123 please can you fix this issue.
Thanks
 
Max

Max

Administrator
Staff member
Hello Muhammad.
Please update to latest WCS build from this page. If the issue still persists, prepare a report as described here and send to support@flashphoner.com with detailed description of your case including SIP accounts to test. We will check.
 
Max

Max

Administrator
Staff member
Hello.
In logs and dump that you provided we cannot see any attempts to send DTMF.
Please clarify, how do you make a SIP call: using WebSDK (if it so, what example do you use?) or REST API? Also, please provide us SIP accounts for testing.
 
M

Muhammad Rizwan

Member
hello,
i didn't understand even single word what you have said please explain in easy words. do you mean SIP account didn't work? if that is the case then i have just test the call using same SIP account that i sent you and its working. please try again.

Thanks
 
Max

Max

Administrator
Staff member
When WCS sends INVITE message to SIP PBX, it responds 200 OK with supported codecs list. In this list, there's no telephone-event codec that is needed to send or receive DTMF signals. It looks that your SIP PBX does not support telephone-event codec and, therefore, does not support DTMF, or this PBX misconfigured.
So, you can:
1. Use another PBX
2. Configure PBX properly
3. Ask PBX technical support to configure it.
 
Max

Max

Administrator
Staff member
Yes, it works with Bria softphone though there is no telephone-event in response SDP. Raised internal issue WCS-1860 for investigation.
 
You must log in or register to reply here.
Top