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> On Behalf Of Tihomir Culjaga Sent: mercoledì 28 febbraio 2018 09:01 To: FreeSWITCH Users Help <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.