[Freeswitch-users] NAT / UDP hole punching issue

Tihomir Culjaga tculjaga at gmail.com
Tue Feb 27 10:50:27 UTC 2018


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="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
SIP-IP                  192.168.100.60
Ext-SIP-IP              85.114.41.180
TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:15061
;maddr=192.168.100.60;transport=tls
WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
WSS-BIND-URL            sips:mod_sofia at 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 at 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 at 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 at 8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
sofia/tls-public/sip:agent2/nexios at 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 at 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 at 8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
sofia/tls-public/sip:agent2/nexios at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 213.147.96.240:10551]
2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
sofia/tls-public/sip:agent2/nexios at 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 at 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 at 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 at 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 at 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 at 213.147.96.240:10551 Standard EXECUTE
EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()



Regards,
Tihomir.
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