This was also my understanding. But as you can understand I want to be certain. Currently this issue only rises with the Cisco hardware in combination with the Barracuda firewall. Other appliances seem to respect the order properly.
On 07-10-16 09:30, Daniel-Constantin Mierla wrote:
I think the Route set as presented in the first email is correct. The Client is also having a proxy, so the caller device has to use the Route set from bottom up, sending first to its proxy, which should send to Kamailio's public IP. If the caller device is sending directly to kamailio, then there is something wrong with that device.
To be complete, this is the entire message:
Via: SIP/2.0/UDP 4.3.2.1:5060;rport=5060;branch=z9hG4bK397b.fc14665.0 Via: SIP/2.0/UDP 10.0.1.50:5060;rport=57093;received=10.0.1.50;branch=z9hG4bK60450F49 Record-Route: sip:192.168.0.200;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes Record-Route: sip:1.2.3.4;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes Record-Route: sip:4.3.2.1;r2=on;lr;did=4c8.3f471377 Record-Route: sip:10.0.0.101;r2=on;lr;did=4c8.3f471377 From: +2233445566 sip:user@sip.domain.net;tag=12D1120C-1C4 To: sip:12345678@sip.domain.net;tag=as70bcd8b2 Call-ID: 6A54E603-894E11E6-8784FA54-A855A3EA@10.0.1.50 CSeq: 102 INVITE Server: sip.domain.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:+2233445566@192.168.0.201:5060 Content-Type: application/sdp Content-Length: 237
v=0 o=charles 2051523742 2051523742 IN IP4 1.2.3.4 s=sip.domain.net c=IN IP4 1.2.3.4 t=0 0 m=audio 13772 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
Cheers, Dirk