WDE Media error 51

Voice & Telephony
1

Hi :slight_smile:

I am using a WebRTC,
I would like to pick up the call on the browser from the WDE (Agent)
What I did is:
A = DN 5000 → Place5000 → WDE:Agent1 on place5000 → Browser uses DN 5000
B= DN 4000 → Place4000 → SipPhone uses DN 4000

I wish that when B calls A, we pick the call from the WDE.
What I get is: when the make the call B - A, the browser is ringing (and I could pick the call) and WDE throw an error: Workspace is requesting a function that is not supported by this T-Server, Media error (51).

I really appreciate your help :slight_smile:
Thank you.

2

Does your DN used by WDE contain the option TServer\sip-cti-control = talk,hold,dtmf ? I believe this is required in order to answer the call from WDE. (Or, at very least, the “talk” value on this option)

1 Like
3

Thank you for your reply.
Yes, they do.
Also, the call from A to B works.

4

You A agent is using WDE with WebRTC, is that it?

5

Yes, exactly.

1 Like
6

Which SIP Endpoint are you using? It must support Broadsoft 3PCC

1 Like
7

Hi.
I dont use sip endpoint. But in WDE option/interaction-workspace, the section about sip endpoint is:

https://drive.google.com/file/d/1L8LQaA74DUGA7RCorNYAKTBMTRlJNVCt/view?usp=sharing

https://drive.google.com/file/d/1L8LQaA74DUGA7RCorNYAKTBMTRlJNVCt/view?usp=sharing
I am curious: is what I am trying to do is possible? ( WDE with Webrtc client (I am using Sipml5)) Or should I use a webPhone.
Thank you

2 Likes
8

??? ???
WebRTC is just a Web protocol, won’t provide RTP alone. SIP can be integrated with WebRTC so it uses the media path it provides.
I suggest you to understand a little bit more on WebRTC architecture, guess you have TURN/STUN servers already deployed?
Sipml5 is just a WebRTC library but as the name suggests, you need SIP.