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(a)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