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SIP/SDP
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Larsen
Brekeke Newbie


Joined: 25 Nov 2017
Posts: 2
Location: Germany

PostPosted: Sat Nov 25, 2017 9:53 am    Post subject: SIP/SDP Reply with quote

1. Brekeke Product Name and Version: 3.7.7.8, Pro Evaluation

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

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Tata
Brekeke Guru


Joined: 27 Jan 2008
Posts: 148

PostPosted: Sun Nov 26, 2017 6:09 pm    Post subject: Reply with quote

Do you want to keep original SDP without Brekeke PBX's modification?

Are there any NAT between both hybrids?
If so, does Brekeke SIP Server need to relay RTP packets without codec changes?
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Larsen
Brekeke Newbie


Joined: 25 Nov 2017
Posts: 2
Location: Germany

PostPosted: Mon Nov 27, 2017 8:17 am    Post subject: Reply with quote

Hello
The thing is, there are hybrids that are connected and sip connected. That goes also the connection is possible with each codec of the hybrid. But then there are still hybrids dial in via a phone number.

+49 211 8894 2234
+49 211 8894 2235
+49 211 8894 2236

The should always be switched to the same number

1034
1035
1036

The numbers are all from a Sipgate trunk. Unfortunately, I can not get it to switch it on. Sad
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Tata
Brekeke Guru


Joined: 27 Jan 2008
Posts: 148

PostPosted: Tue Nov 28, 2017 1:56 pm    Post subject: Reply with quote

Check the SIP Server's log to see INVITE packet sent from Sipgate.
Does a SIP header of INVITE indicate the dial-in number?
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