2 Sep
2014
2 Sep
'14
1:17 p.m.
If you get signling routed ok but no audio, then you have problems
bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use second parameter for
rtpproxy_manage())) to bridge.
I never used sip-natting in kernel, so I am not aware how good it is. I
would rather do just simple port forwarding on the firewall and let
kamailio and rtpproxy to work with advertised address.
Then you can use:
listen=privateip advertise publicip
You will get rid of record routing log message about missing socket.
Otherwise, you should ignore it if the sip routing is ok.
Cheers,
Daniel
On 01/09/14 18:55, Alex Villacís Lasso wrote:
[...]
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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