1. Brekeke Product Name and Version: 3.4.2.1/392
2. Java version: 1.7.0_25
3. OS type and the version: Linux 2.6.32-131.0.15.el6.x86_64
4. UA (phone), gateway or other hardware/software involved: (JsSip (WS) and VoiP Nano (UDP)
5. Your problem:
I´m Testing Brekeke SIP Server with WS Support with JsSIP.
Calls through websocket (WS-WS) progressing with no problems but when the call is from JsSIP to VoIP Nano (WS - UPD) The result is INCOMPATIBLE SDP.
¿Any Idea with this problema? ¿Should I enable any rule in Dial Plan?
Please Help me. Thanks!!
INCOMPATIBLE SDP
Moderator: Brekeke Support Team
Use SIP client which supports ICE.
WebRTC based SIP clients use ICE.
But most non-WebRTC SIP clients don't support ICE.
If your SIP client (on UDP) doesn't support ICE, you need to use a WebRTC gateway.
For example, Brekeke PBX v3.4 can work as a getaway which converts SDP between WebRTC client and non-WebRTC client.
WebRTC based SIP clients use ICE.
But most non-WebRTC SIP clients don't support ICE.
If your SIP client (on UDP) doesn't support ICE, you need to use a WebRTC gateway.
For example, Brekeke PBX v3.4 can work as a getaway which converts SDP between WebRTC client and non-WebRTC client.
You can make a call between WebRTC SIP clients without any issues.
If you want to make a call to non-WebRTC SIP softphone from WebRTC client, a softphone must support ICE, DTLS and SRTP because these protocols are required by WebRTC. (I don't know such a softphone yet..)
Otherwise you need a WebRTC gateway to covert these protocols.
If you want to make a call to non-WebRTC SIP softphone from WebRTC client, a softphone must support ICE, DTLS and SRTP because these protocols are required by WebRTC. (I don't know such a softphone yet..)
Otherwise you need a WebRTC gateway to covert these protocols.