2. Java version: 8
3. OS type and the version:Windows 7
4. UA (phone), gateway or other hardware/software involved: SipTrunk- Sipgate Germany
5. Your problem:
Is it possible to establish a connection via the server / PBX without a codec on the Se server?
Audio Telephone Hybrid -> Brekeke SIP / PBX -> Audio Telephone Hybrid
Both hybrids could be SIP / SDP they are software (LUCI Studio & Luci Live). We actually want the Luci Studio and Luci Live to negotiate the codec they want to use. If we just run the server and call internally from live to studio, we can select all the codecs in Live and the studio routes. If the PBX comes to him from the codec. But we need the PBX to set up the phone numbers and the Sip trunk. Is there a way the SIP / SDP request not on the PBX forward but equal to the terminal?
0exT)-Ek</Z2WINVITE sip:1002@89.244.154.224 SIP/2.0
Via: SIP/2.0/UDP 195.53.0.226:50682;rport;branch=z9hG4bK5449
From: "1003" <sip:1003@89.244.154.224>;tag=28156
To: <sip:1002@89.244.154.224>
Call-ID: 31814
CSeq: 20 INVITE
Contact: "1003" <sip:1003@195.53.0.226:50682>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
Max-Forwards: 70
User-Agent: LuciLiveClient(_3.2.0_)Windows
Subject: This is a N/ACIP call
Content-Length: 356
v=0
o=1003 4717288 0 IN IP4 195.53.0.226
s=LuciLiveClient(_3.2.0_)Windows
c=IN IP4 195.53.0.226
t=0 0
m=audio 5004 RTP/AVP 97
b=TIAS:128000
a=rtpmap:97 mpeg4-generic/48000/1
a=fmtp:97 streamtype=5; profile-level-id=24; config=B98D00; mode=AAC-hbr; sizeLength=13; indexLength=3; indexDeltaLength=3; constantDuration=480; bitrate=128000
a=sendrecv
