Re: [SR-Users] NAT Traversal issue
Dear Kamailio'ns, I am awaiting somebody's suggestions/hints/comments on this issue, with that i can proceed further. Please anybody help me in resolving this issue. Any help will mean a lot and greatly appreciate. Regards, Ravi -- View this message in context: http://sip-router.1086192.n5.nabble.com/NAT-Traversal-issue-tp126442p126456.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] NAT Traversal issue
It looks like you may be running Kamailio behind NAT as well, no? Can you provide any traffic on the connections that fail? Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) On 04/03/2014 08:44 AM, Ravi wrote: Dear Kamailio'ns, I am awaiting somebody's suggestions/hints/comments on this issue, with that i can proceed further. Please anybody help me in resolving this issue. Any help will mean a lot and greatly appreciate. Regards, Ravi ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] NAT Traversal issue
Dear Kamailio'ns,
I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated,
in a standalone intranet infrastructure (no any connection with internet).
I dont have any NAT settings in my network set-up. even though i will not
get Audio/video calls through some times. So in that concern I have
installed RTPproxy, now all the audio/Video calls are fine (with some
Pixelled). I have the Following Kamailio configuration script, in which it
suppose to invoke RTPproxy service when the SIP clients behind NAT. But
every time when i do Audio/Video calls, they are proxying through RTPproxy
server only.
I analysed SIP captures of Audio/video call, i didnt found any IP/port
changes in the whole SIP session and with this i assumed that there is no
NAT issue in my Network.
But why all the Audio/Video calls are proxying through RTPproxy everytime ?
Is there any Wrong placement of function call in Kamailio configuration
script (below) ?
#-
#!ifdef WITH_NAT
# - rtpproxy params -
modparam(rtpproxy, rtpproxy_sock, udp:127.0.0.1:7722)
# - nathelper params -
modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:pinger@192.168.2.52)
modparam(nathelper, sipping_method, INFO)
# - NAT_traversal -
modparam(nat_traversal, keepalive_interval, 60)
modparam(nat_traversal, keepalive_method, NOTIFY)
modparam(nat_traversal, keepalive_state_file,
/var/run/kamailio/keepalive_state)
# - params needed for NAT traversal in other modules -
modparam(nathelper|registrar, received_avp, $avp(RECEIVED))
modparam(usrloc, nat_bflag, 6)
#!endif
#Routing Script
# -
# Sanity Check Section
# -
route {
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
break;
};
#if (msg:len max_len) {
if (msg:len = 8192 ) {
sl_send_reply(513, Message too big);
break;
};
# -
# Record Route Section
# -
#if (method!=REGISTER) {
if (!method==REGISTER) {
record_route();
};
if (method==BYE || method==CANCEL) {
#unforce_rtp_proxy();
rtpproxy_answer();
}
# -
# Loose Route Section
# -
if (loose_route()) {
if ((method==INVITE || method==REFER) !has_totag()) {
sl_send_reply(403, Forbidden);
break;
};
if (method==INVITE) {
if (!proxy_authorize(192.168.2.52,subscriber)) {
#proxy_challenge(,0);
proxy_challenge(192.168.2.52, 0);
#break;
}
else if (!check_from()) {
sl_send_reply(403, Use From=ID);
break;
};
consume_credentials();
if (nat_uac_test(19)) {
setflag(6);
force_rport();
fix_nated_contact();
};
rtpproxy_offer(l);
};
route(1);
break;
};
# -
# Call Type Processing Section
# -
#if (uri!=myself) {
if (!uri==myself) {
route(4);
route(1);
break;
};
if (method==ACK) {
route(1);
break;
}
if (method==CANCEL) {
route(1);
break;
}
else if (method==INVITE) {
route(3);
break;
}
else if (method==REGISTER) {
route(2);
break;
};
lookup(aliases);
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup(location)) {
sl_send_reply(404, User Not Found);
break;
};
route(1);
}
# -
# Default Message Handler
# -
route[1] {
t_on_reply(1);
if (!t_relay()) {
if (method==INVITE isflagset(6)) {
rtpproxy_answer();;
};
sl_reply_error();
};
}
# -
# REGISTER Message Handler
# -
route[2] {
if (!search(^Contact:[ ]*\*) nat_uac_test(19)) {
setflag(6);
fix_nated_register();
force_rport();
};
sl_send_reply(100, Trying);
if (!www_authorize(192.168.2.52,subscriber)) {
www_challenge(192.168.2.52,0);
break;
};
if (!check_to()) {
sl_send_reply(401, Unauthorized);
break;
};
consume_credentials();
if (!save(location)) {
sl_reply_error();
};
}
# -
# INVITE Message
Re: [SR-Users] NAT Traversal issue
Hello, great, thanks for replying and closing the thread with the solution. Cheers, Daniel On 12/19/12 2:31 AM, Raj Roy Ghandhi wrote: Hi All, Problem solved. It was a CODEC issue. Best Regards, Roy. On Tue, Dec 4, 2012 at 3:58 AM, Raj Roy Ghandhi roy.gan...@gmail.com mailto:roy.gan...@gmail.com wrote: Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client-LAN client web page call 2. LAN webrtc client - 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] NAT Traversal issue
Hi All, Problem solved. It was a CODEC issue. Best Regards, Roy. On Tue, Dec 4, 2012 at 3:58 AM, Raj Roy Ghandhi roy.gan...@gmail.comwrote: Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client-LAN client web page call 2. LAN webrtc client - 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] NAT Traversal issue
Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client-LAN client web page call 2. LAN webrtc client - 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy. interface: eth0 (50.62.131.0/255.255.255.0) match: port=5060 T 50.62.131.23:8080 - 103.247.48.130:55161 [AP] .~..ACK sip:1004@103.247.48.130:55161;transport=ws SIP/2.0. Via: SIP/2.0/WS 50.62.131.23;branch=z9hG4bK360e.00026edd70b96704cb81c80a68974567.0. Via: SIP/2.0/UDP 50.62.131.23;rport=5060;branch=z9hG4bK360e.f4821c8cbfe17519eab77c646bac3b28.0. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=112.135.253.137;branch=z9hG4bKOabKiBFSHWoaW3Tlfxnl;rport=10921. From: sip:1005@50.62.131.23;tag=HR3zr1RB3pFAyiCMroCb. To: sip:1004@50.62.131.23;tag=vAEr1XA08CEPeSeQyNZP. Contact: allansip:1005@112.135.253.137:10921;transport=ws;alias=112.135.253.137~10921~5;+sip.ice. Call-ID: b92e2549-e075-c2dc-4cd7-15d0453dbead. CSeq: 50914 ACK. Content-Length: 0. Max-Forwards: 68. Proxy-Authorization: Digest username=1005,realm=50.62.131.23,nonce=UL1/4FC9frSo6PhNP9NrXpGTvo6il6fD,uri=sip:1004@103.247.48.130:55161;transport=ws;alias=103.247.48.130~55161~5,response=7a17d49ef95a2032bb9d95c5fb8fd959,algorithm=MD5. User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/. Organization: Stellooo Telecom. . T 50.62.131.23:8080 - 103.247.48.130:55161 [A] .~..INVITE sip:1004@103.247.48.130:55161;transport=ws SIP/2.0. Via: SIP/2.0/WS 50.62.131.23;branch=z9hG4bK460e.875dc75862ecb1152fcf8b8a1688bb1f.0. Via: SIP/2.0/UDP 50.62.131.23;rport=5060;branch=z9hG4bK460e.d4b479eb2a5d0080a4b667a62e5b6e11.0. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=112.135.253.137;branch=z9hG4bKgLHVpHaMKJtI3PIB7P3dKET9rAMz88p7;rport=10921. From: sip:1005@50.62.131.23;tag=HR3zr1RB3pFAyiCMroCb. To: sip:1004@50.62.131.23;tag=vAEr1XA08CEPeSeQyNZP. Contact: allansip:1005@112.135.253.137:10921;transport=ws;alias=112.135.253.137~10921~5;+sip.ice. Call-ID: b92e2549-e075-c2dc-4cd7-15d0453dbead. CSeq: 50915 INVITE. Content-Type: application/sdp. Content-Length: 2009. Max-Forwards: 68. Proxy-Authorization: Digest username=1005,realm=50.62.131.23,nonce=UL1/4FC9frSo6PhNP9NrXpGTvo6il6fD,uri=sip:1004@103.247.48.130:55161;transport=ws;alias=103.247.48.130~55161~5,response=83a36fa696550fdbbab901764d50b810,algorithm=MD5. User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/. Organization: Stellooo Telecom. . v=0. o=- 1778509583 3 IN IP4 50.62.131.23. s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000). t=0 0. a=group:BUNDLE audio video. m=audio 60846 RTP/SAVPF 103 104 0 8 106 105 13 126. c=IN IP4 50.62.131.23. a=rtcp:60847. a=candidate:2205430794 1 udp 2113937151 192.168.1.25 53592 typ host generation 0. a=candidate:2205430794 2 udp 2113937151 192.168.1.25 53592 typ host generation 0. a=candidate: T 50.62.131.23:8080 - 103.247.48.130:55161 [AP] .~..ACK sip:1004@103.247.48.130:55161;transport=ws SIP/2.0. Via: SIP/2.0/WS 50.62.131.23;branch=z9hG4bK360e.bf139f85992792dc89d292461931f9c0.0. Via: SIP/2.0/UDP 50.62.131.23;rport=5060;branch=z9hG4bK360e.9cd3a5cf9d235604940851528556017d.0. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=112.135.253.137;branch=z9hG4bKh023sH2ZTwcU44wzK5JP;rport=10921. From: sip:1005@50.62.131.23;tag=HR3zr1RB3pFAyiCMroCb. To: sip:1004@50.62.131.23;tag=vAEr1XA08CEPeSeQyNZP. Contact: allansip:1005@112.135.253.137:10921;transport=ws;alias=112.135.253.137~10921~5;+sip.ice. Call-ID: b92e2549-e075-c2dc-4cd7-15d0453dbead. CSeq: 50914 ACK. Content-Length: 0. Max-Forwards: 68. Proxy-Authorization: Digest username=1005,realm=50.62.131.23,nonce=UL1/4FC9frSo6PhNP9NrXpGTvo6il6fD,uri=sip:1004@103.247.48.130:55161;transport=ws;alias=103.247.48.130~55161~5,response=7a17d49ef95a2032bb9d95c5fb8fd959,algorithm=MD5. User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/. Organization: Stellooo Telecom. . T 50.62.131.23:8080 - 112.135.253.137:10921 [AP] .~..BYE sip:1005@112.135.253.137:10921;transport=ws SIP/2.0. Via: SIP/2.0/WS 50.62.131.23;branch=z9hG4bKcde9.dc2be89cbace6ff6bbc29682239b388a.0. Via: SIP/2.0/UDP 50.62.131.23;rport=5060;branch=z9hG4bKcde9.586031647d4f56fb6d9645073c0599ff.0. Via: SIP/2.0/WS df7jal23ls0d.invalid;received=103.247.48.130;branch=z9hG4bKZ1gAmd5aaMk8RXx9tpznbqMiTdT8mmJs;rport=55161. From: sip:1004@50.62.131.23;tag=vAEr1XA08CEPeSeQyNZP. To: sip:1005@50.62.131.23;tag=HR3zr1RB3pFAyiCMroCb. Call-ID: b92e2549-e075-c2dc-4cd7-15d0453dbead. CSeq: 34113 BYE. Content-Length: 0. Max-Forwards: 68. User-Agent:
