[Freeswitch-users] NAT / UDP hole punching issue

Wed Feb 28 08:18:07 UTC 2018

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 at lists.freeswitch.org> On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:01
To: FreeSWITCH Users Help <freeswitch-users at 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 at gmail.com <mailto:tculjaga at 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 at 85.114.41.180:15061 <http://sip:mod_sofia@85.114.41.180:15061> 

TLS-BIND-URL            sips:mod_sofia at 85.114.41.180 <mailto:sips%3Amod_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 <mailto:sip%3Amod_sofia at 192.168.100.60> :5066;transport=ws

WSS-BIND-URL            sips:mod_sofia at 192.168.100.60 <mailto:sips%3Amod_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 <mailto: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 <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 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 <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 at 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 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 <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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 213.147.96.240:10551 <http://nexios@213.147.96.240:10551>  Standard EXECUTE

EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios@213.147.96.240:10551>  echo()

Regards,

Tihomir.

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