[...]
Maybe I should explain my setup better.
The test setup I want to run is, in a way, twice natted. The asterisk instance
runs in localhost, the innermost net. The asterisk is not supposed to get its
SIP signaling from anyone but Kamailio. The /etc/asterisk/sip.conf contains
this:
[root@elx3 ~]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
allowguest=no
realm=asterisk
srvlookup=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
relaxdtmf=yes
trustrpid=no
sendrpid=yes
sendrpid=pai
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
faxdetect=yes
t38pt_udptl=yes
vmexten=*97
videosupport=yes
maxcallbitrate=384
nat=force_rport,comedia
directmedia=no
accept_outofcall_message=yes
auth_message_requests=yes
;The following settings restrict Asterisk to localhost for Kamailio integration
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
bindport=5080
outboundproxy=127.0.0.1
outboundproxyport=5060
#include sip_general_custom.conf
#include sip_register.conf
#include sip_custom.conf
The kamailio instance is the first instance of routing. It listens on all interfaces on port 5060 and routes packets from all the other interfaces to localhost and back. One of these interfaces is the local network (192.168.2.18), which routes to our
gateway.
If kamailio is given a public interface, then our setup works correctly. The exchange in the first mail shows what happens when the packet is routed through our gateway (the second instance of routing, and an actual NAT). Our gateway is a linux system
with a kernel module (nf_nat_sip, nf_conntrack_sip) that rewrites the headers on the fly, resulting in the packet exchange as seen in the first mail. From what I have seen, the kernel modules rewrite To, From, but not Record-Route, where an instance of
the internal IP remains. If I understand correctly, the remote system tries to route to its own interpretation of 192.168.2.18, which fails.
If I add the advertised_address parameter and set it to the public IP, outgoing
calls from asterisk to a registered SIP client break and get established with
no audio (tested with Jitsi). I get the following exchange from 192.168.2.18:
INVITE
sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com
SIP/2.0
Record-Route: <sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Max-Forwards: 69
From: "Anonymous" <sip:anonymous@anonymous.invalid:5080>;tag=as0551c44f
To:
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>
Contact: <sip:anonymous@127.0.0.1:5080>
Call-ID: 4f40ffcc123459313daf47397e18b0af@127.0.0.1:5080
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Mon, 01 Sep 2014 16:11:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
v=0
o=root 1851320733 1851320733 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 18624 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
To:
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP
127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route:
<sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Call-ID: 4f40ffcc123459313daf47397e18b0af@127.0.0.1:5080
From: "Anonymous" <sip:anonymous@anonymous.invalid:5080>;tag=as0551c44f
Contact: "avillacis"
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>
User-Agent: Jitsi2.5.5255Linux
Content-Length: 0
SIP/2.0 200 OK
To:
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.6bce295bab666c7aceeddfebdc70c190.0;received=192.168.2.18,SIP/2.0/UDP
127.0.0.1:5080;branch=z9hG4bK696d3bf6;rport=5080
Record-Route:
<sip:201.234.196.170;r2=on;lr=on;ftag=as0551c44f>,<sip:127.0.0.1;r2=on;lr=on;ftag=as0551c44f>
CSeq: 102 INVITE
Call-ID: 4f40ffcc123459313daf47397e18b0af@127.0.0.1:5080
From: "Anonymous" <sip:anonymous@anonymous.invalid:5080>;tag=as0551c44f
Contact: "avillacis"
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>
User-Agent: Jitsi2.5.5255Linux
Content-Type: application/sdp
Content-Length: 217
v=0
o=avillacis-jitsi.org 0 0 IN IP4 192.168.3.2
s=-
c=IN IP4 192.168.3.2
t=0 0
m=audio 5006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
ACK
sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com
SIP/2.0
Via: SIP/2.0/UDP
201.234.196.170;branch=z9hG4bKd5f3.805fac86b0d1e9c1dc577d5ca12f12d3.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2d32b799;rport=5080
Max-Forwards: 69
From: "Anonymous" <sip:anonymous@anonymous.invalid:5080>;tag=as0551c44f
To:
<sip:avillacis@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=d28a3a6e
Contact: <sip:anonymous@127.0.0.1:5080>
Call-ID: 4f40ffcc123459313daf47397e18b0af@127.0.0.1:5080
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.12.0
Content-Length: 0
At the same time, I get this on the kamailio log:
WARNING: rr [loose.c:830]: after_loose(): no socket found for match second RR
If I try the incoming call from the internet, while advertised_address is
enabled, I get the following exchange. I also get the exact same log message,
and one-way audio.
INVITE sip:*43@201.234.196.170:5060 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport
Max-Forwards: 70
From: "9003" <sip:9003@198.58.101.75>;tag=as69ee0744
To: <sip:*43@201.234.196.170:5060>
Contact: <sip:9003@198.58.101.75:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe@198.58.101.75:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Mon, 01 Sep 2014 16:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301
v=0
o=root 1281221163 1281221163 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 12958 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
From: "9003" <sip:9003@198.58.101.75>;tag=as69ee0744
To: <sip:*43@201.234.196.170:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe@198.58.101.75:5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK587b52bc;rport=5060
Record-Route:
<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Record-Route:
<sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
From: "9003" <sip:9003@198.58.101.75>;tag=as69ee0744
To: <sip:*43@201.234.196.170:5060>;tag=as5f3239b9
Call-ID: 0398a11d3149031240ec2e70077a99fe@198.58.101.75:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*43@127.0.0.1:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 277
v=0
o=root 1757515753 1757515753 IN IP4 127.0.0.1
s=Asterisk PBX 11.12.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 16396 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
ACK sip:*43@127.0.0.1:5080 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK636e2948;rport
Route:
<sip:192.168.2.18;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->,<sip:201.234.196.170;r2=on;lr=on;ftag=as69ee0744;vsf=SBoZSkpbSEZaLF1YW0dGeB8ICB8bDxsxMDEuNzU->
Max-Forwards: 70
From: "9003" <sip:9003@198.58.101.75>;tag=as69ee0744
To: <sip:*43@201.234.196.170:5060>;tag=as5f3239b9
Contact: <sip:9003@198.58.101.75:5060>
Call-ID: 0398a11d3149031240ec2e70077a99fe@198.58.101.75:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0
How can I fix the "no socket found for match second RR" error?
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