Sorry to hear Tihomir. Is there ANYWAY you can test everything locally on same network (i.e. without stun, nat, etc..) maybe the issue isn’t with stun! From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of Tihomir Culjaga Sent: mercoledì 28 febbraio 2018 09:57 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue RTP ports are defined. When i do a port forward for my RTP range i get my RTP audio working i guess due to rtp-auto-adjust feature on FS... but it should work without port forwarding. here simply FS is not starting to send RTP traffic to the client even if it notified its public IP:PORT in SDP on 200 OK. i see FS contacting a STUN server, getting the public IP:PORT and than ... doesn't send any RTP traffic towards the client... this is what its bugging me. T. On 28 February 2018 at 09:18, Francesco Facco de Lagarda <francesco@delagarda.com <mailto:francesco@delagarda.com> > wrote: Check your RTP ports .. in the fs config and the port forwarding on firewalls. Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know! Good luck! F From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > On Behalf Of Tihomir Culjaga Sent: mercoledì 28 febbraio 2018 09:01 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue does anyone have a clue ? :=) On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga@gmail.com <mailto:tculjaga@gmail.com> > wrote: hi, I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :( my setup is like this: Phone <> NAT <> INTERNET <> NAT <FreeSWITCH> FreeSWITCH version: 1.6.12~64bit ( 64bit) I have a separate profile configured for TLS: <param name="rtp-ip" value="192.168.100.60"/> <param name="sip-ip" value="192.168.100.60"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="ext-sip-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="tls-only" value="true"/> <param name="tls-sip-port" value="15061"/> ================================================================================================= Name tls-public Domain Name N/A Auto-NAT false DBName sofia_reg_tls-public Pres Hosts 192.168.100.60,192.168.100.60 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.100.60 Ext-RTP-IP stun:stun.freeswitch.org <http://stun.freeswitch.org> SIP-IP 192.168.100.60 Ext-SIP-IP 85.114.41.180 TLS-URL sip:mod_sofia@85.114.41.180:15061 <http://sip:mod_sofia@85.114.41.180:15061> TLS-BIND-URL sips:mod_sofia@85.114.41.180 <mailto:sips%3Amod_sofia@85.114.41.180> :15061;maddr=192.168.100.60;transport=tls WS-BIND-URL sip:mod_sofia@192.168.100.60 <mailto:sip%3Amod_sofia@192.168.100.60> :5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.100.60 <mailto:sips%3Amod_sofia@192.168.100.60> :7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA CODECS OUT PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 0 i manage to register the phone with no problems but when i call the phone i get no audio; bgapi expand originate ${sofia_contact(tls-profile/agent2/nexios@192.168.100.60 <mailto:nexios@192.168.100.60> )} &echo() FS sends the invite as: SDP in INVITE message from FS v=0 o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180 s=FreeSWITCH c=IN IP4 85.114.41.180 t=0 0 m=audio 17480 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 SIP Client responds with: SDP in 200 OK from the client v=0 o=- 3728718779 3728718780 IN IP4 213.147.96.240 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 8 101 c=IN IP4 213.147.96.240 b=TIAS:64000 a=rtcp:4003 IN IP4 213.147.96.240 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the UDP stream is: client( 4002 ) <> ( 17480 )FS when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client. so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ? in FS logs i see 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> entering state [ready][200] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Original read codec set to PCMA:8 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Set 2833 dtmf send payload to 101 recv payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 20 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 <http://213.147.96.240:4003> 2 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Set 2833 dtmf send payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Set 2833 dtmf receive payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Set rtp dtmf delay to 40 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ] has been answered 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ) Callstate Change RINGING -> ACTIVE 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ] 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Flipping CID from "" <0000000000> to "Outbound Call" <nexios> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ) Running State Change CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> ) State EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> SOFIA EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> Standard EXECUTE EXECUTE sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 <http://nexios@213.147.96.240:10551> echo() Regards, Tihomir. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 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