On 28 February 2018 at 09:18, Francesco Facco de Lagarda <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> 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> 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"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>



<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
SIP-IP 192.168.100.60
Ext-SIP-IP 85.114.41.180
TLS-URL sip:mod_sofia@85.114.41.180:15061
TLS-BIND-URL sips:mod_sofia@85.114.41.180:15061;maddr=192.168.100.60;transport=tls
WS-BIND-URL sip:mod_sofia@192.168.100.60:5066;transport=ws
WSS-BIND-URL sips:mod_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)} &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 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 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 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 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] 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 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 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 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 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] 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) 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]
2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/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) 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) 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) 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 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 Standard EXECUTE
EXECUTE sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 echo()

Regards,
Tihomir.

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