Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

El 01/09/14 10:50, Alex Villací­s Lasso escribió: 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?