Re: [SR-Users] fix_nated_sdp issue
[parser/sdp/sdp_helpr_funcs.c:479]: located IP address [10.0.10.11] in `c=' field Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy [rtpproxy.c:2237]: proxy reply: 38946 10.0.10.10#012 Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: siputils [checks.c:104]: no totag Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: core [msg_translator.c:457]: clen_builder: content-length: 347 (347) Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: core [msg_translator.c:204]: check_via_address(10.0.10.11, 10.0.10.11, 0) Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: script: -- exiting relaying --- Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: core [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Walking through the log makes me think that because I'm using rtpproxy and nathelper, when the t_relay fires it errantly appends the address for rtpproxy to the c= line... Am I going about this all wrong - is there a better approach? you cannot use manage_rtpproxy (or other functions from rtpproxy module updating the sdp) with fix_nated_sdp() because of the way changes are applied to the sip message. When using both, it results in concatenation of the two IP -- it is why I asked about the log, expecting you used such two functions. You may try using msg_apply_changes() in between such two functions, but I recommend making the config file decision in a way that you execute only one such function. For example, you can set a flag after using such functions and before using another one test that flag. Cheers, Daniel Ric *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com] *Sent:* Wednesday, February 22, 2012 12:52 AM *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Cc:* Ric Marques *Subject:* Re: [SR-Users] fix_nated_sdp issue Hello, can you set debug=3 in the config file and send the output (syslog messages) of processing such invite? Cheers, Daniel On 2/22/12 4:31 AM, Ric Marques wrote: Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider - nat firewall - kamailio/rtpproxy - asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp(2, 10.50.50.8) to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -- INVITE sip:19165551...@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: 1009 sip:1009@10.0.10.11;tag=as5498b77e To: sip:19165551...@xxx.xxx.xxx.xxx Contact: sip:1009@10.0.10.11:5060 Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 mailto:06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -- Here's the sip invite with the 'fix_nated_sdp' command: -- INVITE sip:19167828...@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch
[SR-Users] fix_nated_sdp issue
Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider - nat firewall - kamailio/rtpproxy - asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp(2, 10.50.50.8) to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -- INVITE sip:19165551...@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: 1009 sip:1009@10.0.10.11;tag=as5498b77e To: sip:19165551...@xxx.xxx.xxx.xxx Contact: sip:1009@10.0.10.11:5060 Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -- Here's the sip invite with the 'fix_nated_sdp' command: -- INVITE sip:19167828...@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060 Max-Forwards: 69 From: 1009 sip:1009@10.0.10.11;tag=as49e00c81 To: sip:19167828...@xxx.xxx.xxx.xxx Contact: sip:1009@10.0.10.11:5060 Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 P-hint: outbound v=0 o=root 1009117068 1009117068 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.50.50.8.10.0.10.10 t=0 0 m=audio 13540 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=oldmediaip:10.0.10.11 a=nortpproxy:yes -- Is this a bug, or is it likely I have something else screwed up? Thank you in advance for your assistance - this list is an incredible resource! -Ric ___ 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] fix_nated_sdp()
Hi Bernhard, have you checked the SDP which is going to and from the Freeswitch-Server? Is it modified? If yes: You may want to check your NAT-Firewall. A working (but not nice) solution might be to relay the RTP through an proxy in this case... If No: Verify the result of nat_uac_test: Why is it returning false? Carsten 2011/1/25 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, someone has an idea? Thanks in advance! Best regards, Bernhard -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Bernhard Suttner Gesendet: Montag, 24. Januar 2011 20:38 An: sr-users@lists.sip-router.org Betreff: [SR-Users] fix_nated_sdp() Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ 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 ___ 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 -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Büro (Verl) +49 5246 801427 Fax +49 40 34927218 mailto:cars...@bock.info ___ 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] fix_nated_sdp()
Hi, thanks for your answer. The question was about, if the fix_nated_sdp() (in the example below) could maybe break something. I am sure, that the fix_nated_sdp() would work in some cases but I am not sure, if the function could break something: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Or is there a better solution to fix these damend NAT scenarios? Best regards, Bernhard -Ursprüngliche Nachricht- Von: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Im Auftrag von Carsten Bock Gesendet: Dienstag, 25. Januar 2011 18:56 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] fix_nated_sdp() Hi Bernhard, have you checked the SDP which is going to and from the Freeswitch-Server? Is it modified? If yes: You may want to check your NAT-Firewall. A working (but not nice) solution might be to relay the RTP through an proxy in this case... If No: Verify the result of nat_uac_test: Why is it returning false? Carsten 2011/1/25 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, someone has an idea? Thanks in advance! Best regards, Bernhard -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Bernhard Suttner Gesendet: Montag, 24. Januar 2011 20:38 An: sr-users@lists.sip-router.org Betreff: [SR-Users] fix_nated_sdp() Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ 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 ___ 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 -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Büro (Verl) +49 5246 801427 Fax +49 40 34927218 mailto:cars...@bock.info ___ 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] fix_nated_sdp()
Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard - Original Message - From: Ovidiu Sas [mailto:o...@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: Carsten Bock [mailto:li...@bock.info], sr-users@lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() You need to experiment with nathelper module and while doing that, you need to inspect the SIP messages in order to understand how it works. Only when you fully understand your scenarios, you will be able to properly configure, debg and setup kamailio. Best thing to do here is to use rtpproxy and relay media if needed. Another option would be to let the SDP untouched and let freeswitch do it's magic (if it can). Not sure if this will work in all scenarios. Calling fix_nated_sdp() on the SBC will definitely not work (you are just messing with the IP on the SDP and you don't know if the real RTP port on the NATed side will match the advertised port in your fixed SDP). Regards, Ovidiu Sas On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner bernhard.sutt...@winet.ch wrote: Hi, thanks for your answer. The question was about, if the fix_nated_sdp() (in the example below) could maybe break something. I am sure, that the fix_nated_sdp() would work in some cases but I am not sure, if the function could break something: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Or is there a better solution to fix these damend NAT scenarios? Best regards, Bernhard -Ursprüngliche Nachricht- Von: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Im Auftrag von Carsten Bock Gesendet: Dienstag, 25. Januar 2011 18:56 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] fix_nated_sdp() Hi Bernhard, have you checked the SDP which is going to and from the Freeswitch-Server? Is it modified? If yes: You may want to check your NAT-Firewall. A working (but not nice) solution might be to relay the RTP through an proxy in this case... If No: Verify the result of nat_uac_test: Why is it returning false? Carsten 2011/1/25 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, someone has an idea? Thanks in advance! Best regards, Bernhard -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Bernhard Suttner Gesendet: Montag, 24. Januar 2011 20:38 An: sr-users@lists.sip-router.org Betreff: [SR-Users] fix_nated_sdp() Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip
[SR-Users] fix_nated_sdp()
Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ 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
