30 Apr
2010
30 Apr
'10
11:10 p.m.
Daniel,
Maybe my question is silly, but in this case the contact should remain
intact? (I mean in bridge mode).
I understand the destination UA should read the Record-route Headers and
ignore the contents of the Contact Header, but I think this is not what´s
happening.
I'm not using force_socket because the gateway already knows how to route
the calls and I'm detecting the outgoing interface before calling
force_rtp_proxy with flags.
Should I replace the contact using REGEX?
Thanks,
Uriel
On Fri, Apr 30, 2010 at 3:03 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
Hello,
it might not be the solution, because they should route based on
Record-Route headers, not on Contact header. Anyhow changing the Contact
will break the routing, so you will need to store somehow the original
contact.
You can do manual detection in case you do bridging, by checking the
receiving interface, $Ri is the local IP where the request was received,
therfore you will be sending on the other interface. Are you doing force
send socket to select outgoing interface? If yes, then is where you know the
local ip for sending.
Cheers,
Daniel
On 4/30/10 6:32 PM, Uriel Rozenbaum wrote:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most
of my Gateways. My setup includes two different interfaces one with a public
IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls (or
calls will have some strange behavior) if the Contact header has an IP out
of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already the
one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X><sip:60911100@192.168.200.X>
;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y> <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X><sip:60911100@192.168.200.X>
;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y> <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using and fix
the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
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Daniel-Constantin Mierla
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