19 Feb
2013
19 Feb
'13
10:20 a.m.
Hi,
When kamailio is listening on more than one interface is is sometimes
necessary to use force_send_socket() to tell kamailio to use one socket
or the other. This means it will put the correct values into Via and
Record-Route headers. In the example below, I am choosing the socket
based on if the request has come from a dispatcher list, but any other
logic can be used.
#!substdef "!WAN_IP!x.x.x.x!g"
#!substdef "!LAN_IP!y.y.y.y!g"
route[RELAY]{
if (ds_is_from_list("CLUSTER_ID")) {
xlog("L_INFO", "Using WAN_IP for sending\n");
force_send_socket("WAN_IP":5060);
} else {
xlog("L_INFO", "Using LAN_IP for sending\n");
force_send_socket("LAN_IP":5060);
}
t_relay();
}
Regards,
Hugh
On 18/02/2013 18:28, Scott, Matt wrote:
I can get wan communication or lan communication
working, but not both.
Setting this, allows wan communication, but then my lan tries to talk with my wan ip.
Single Interface: listen=udp:<lan_ip>:5060 advertise <nat_wan_ip>:5060
I see my wan_ip in the via header, so I assume that's causing the issues for my lan.
Using add_contact_alias for lan, with the advertise in my listen command, results in
[nathelper.c:835]: no need to add alias param
I've tried handle_ruri_alias();, but this doesn't seem to have an affect either.
Setting this, allows lan communication to work fine, but wan has issues.
Single Interface: listen=udp:<lan_ip>:5060
Where is the happy median?
Was going to try path, but I read this is not the correct work-around.
Any clues or hints, are greatly appreciated.
U <carrier_wan_ip>:5060 -> <lan_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Max-Forwards: 63..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept:
application/sdp, appl
ication/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1720276
3205@<carrier_wan_ip:5060>..P-Asserted-Identity:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100
rel..Content-Length: 305..Content-Disposition: session;
handling=required..Content-Type: applic
ation/sdp....v=0..o=Sonus_UAC 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media
Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0
PCMU/8000..a=rtpmap:8 PCMA/800
0..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-1
5..a=sendrecv..a=maxptime:20..
U <lan_ip>:5060 -> <carrier_wan_ip>:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
<carrier_wan_ip:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Server: kamailio (3.3.2 (x86_64/linux))..Content-Length: 0....
U <lan_ip>:5060 -> <asterisk_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<wan_ip>:5060;branch=
z9hG4bK5d66.7cb4b7a1.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3..Fro
m:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To:
<sip:+1<dialed_number>@192.168.
9.130:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip>..CSeq: 27475
INVITE..Max-Forwards: 62..Allo
w: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup,
application/dt
mf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..P-As
serted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported:
100rel..Content-Length: 305..C
ontent-Disposition: session; handling=required..Content-Type:
application/sdp....v=0..o=Sonus_UA
C 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio
15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18
G729/8000..a=
fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=sendrecv..a=maxptime:20
..
Asterisk box:
Retransmitting #4 (no NAT) to <wan_ip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
<wan_ip>:5060;branch=z9hG4bK5d66.7cb4b7a1.0;received=192.168.9.130
Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3
From:
<sip:+1<caller_id>@<carrier_wan_ip>:5060;isup-oli=0>;tag=gK0413c49b
To: <sip:+1<dialed_number>@<lan_ip>:5060>;tag=as1265f4d0
Call-ID: 184873373_18184370@<carrier_wan_ip>
CSeq: 27475 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:+1<dialed_number>@<asterisk_ip>>
Content-Type: application/sdp
Content-Length: 239
Matt Scott
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Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.