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Hi all,</div>
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I am setting up a proxy to route calls between a cloud 3CX instance and MS Teams Direct Routing. At the moment I'm just trying to get the OpenSIPS to 3CX connection working, testing with a softphone pointed at the proxy and attempting to make calls to extensions
on the 3CX.</div>
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Opensips is behind NAT with two sockets configured, one TCP with an "as" setting for the public address of the NAT, one UDP which is where the softphone connects. Outbound calls from the softphone as UAC to 3CX extensions as UAS ring, show established when
answered, and I get audio at least one way (I think the one way audio may be a firewall issue). There are no registrations being used by any endpoint.<br>
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Opensips adds a double record-route header as expected for both the receiving and sending interface.<br>
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The problem I am having is that a BYE from one of the extensions on the 3CX (being a callee/UAS) arrives at OpenSIPs on the public-facing socket, with a double route header, in the order of public, private, with lr and r2=on set, but loose_route() does not
behave according to the RFCs. I believe it should remote both route headers and then send the BYE to the destination in the RURI, spiralling back into the proxy via the sip address in the last Route: header and then being routed to the softphone/UAC.<br>
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Instead, it seems to replace the RURI with the sip address of the last Route header, and sends the message via TCP to the internal socket (which I therefore had to add TCP to - if I dont have both UDP and TCP on the internal socket, we get a failed TCP connection
here), thus causing a loop and an eventual Too Many Hops.</div>
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My openSIPS config is here:</div>
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<a href="https://pastebin.com/4t006vfn" id="LPlnk">https://pastebin.com/4t006vfn</a></div>
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A packet capture of the BYE and subsequent problem, the public facing private ip is 10.20.12.124 (TCP socket), advertised as 195.196.197.198, and the internal private IP (facing the softphone) is 10.20.12.125 with a TCP and a UDP socket. 3CX is 62.63.64.65.<span style="font-size: 12pt; background-color: rgb(51, 51, 51); color: rgb(0, 0, 0);"></span><span style="font-size: 12pt; background-color: rgb(51, 51, 51); color: rgb(0, 0, 0);"></span><span style="font-size: 12pt; background-color: rgb(51, 51, 51); color: rgb(0, 0, 0);"></span><span style="font-size: 12pt; background-color: rgb(51, 51, 51); color: rgb(0, 0, 0);"></span>
Softphone (UAC) uses UDP, 3CX uses TCP. SIP Domain of the proxy and softphone is 195.196.197.198, the softphone's SIP username is 1838, I'm calling extension 5760 on the 3CX.<br>
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BYE from 3XC (from UAS)<br>
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Session Initiation Protocol (BYE)
<div> Request-Line: BYE sip:1838@195.196.197.198:5060;transport=tcp SIP/2.0</div>
<div> Message Header</div>
<div> Via: SIP/2.0/TCP 62.63.64.65:5060;branch=z9hG4bK-524287-1---77636e709985902e;rport</div>
<div> Max-Forwards: 70</div>
<div> Route: <sip:195.196.197.198:5060;transport=tcp;lr;r2=on;ftag=lihnv;nat=yes></div>
<div> Route: <sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes></div>
<div> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></div>
<div> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</div>
<div> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</div>
<div> Call-ID: mpytwgabkcpnkef@ryzing</div>
<div> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</div>
<div> CSeq: 2 BYE</div>
<div> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</div>
<span> Content-Length: 0</span></div>
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<span>BYE after loose_route():<br>
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<span></span>Session Initiation Protocol (BYE)
<div> Request-Line: BYE sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes SIP/2.0</div>
<div> Message Header</div>
<div> Via: SIP/2.0/TCP 195.196.197.198:5060;branch=z9hG4bKcd21.3682afa6.0;i=a10f2936</div>
<div> Via: SIP/2.0/TCP 62.63.64.65:5060;received=62.63.64.65;branch=z9hG4bK-524287-1---77636e709985902e;rport=44953</div>
<div> Max-Forwards: 69</div>
<div> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></div>
<div> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</div>
<div> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</div>
<div> Call-ID: mpytwgabkcpnkef@ryzing</div>
<div> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</div>
<div> CSeq: 2 BYE</div>
<div> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</div>
<span> Content-Length: 0</span></div>
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Then this just loops around as expected due to the RURI in the above message:</div>
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Session Initiation Protocol (BYE)
<div> Request-Line: BYE sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes SIP/2.0</div>
<div> Message Header</div>
<div> Via: SIP/2.0/UDP 10.20.12.125:5060;branch=z9hG4bKcd21.4682afa6.0;i=c10f2936</div>
<div> Via: SIP/2.0/TCP 195.196.197.198:5060;rport=47455;received=10.20.12.124;branch=z9hG4bKcd21.3682afa6.0;i=a10f2936</div>
<div> Via: SIP/2.0/TCP 62.63.64.65:5060;received=62.63.64.65;branch=z9hG4bK-524287-1---77636e709985902e;rport=44953</div>
<div> Max-Forwards: 68</div>
<div> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></div>
<div> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</div>
<div> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</div>
<div> Call-ID: mpytwgabkcpnkef@ryzing</div>
<div> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</div>
<div> CSeq: 2 BYE</div>
<div> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</div>
<span> Content-Length: 0</span></div>
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<span>Any clues or ways of resolving this would be very much appreciated!</span></div>
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<span>Best regards</span></div>
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<span>Alex</span><br>
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