[Freeswitch-users] NAT / UDP hole punching issue

Wed Feb 28 09:34:44 UTC 2018

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 at lists.freeswitch.org> On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:57
To: FreeSWITCH Users Help <freeswitch-users at 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 at delagarda.com <mailto:francesco at 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 at lists.freeswitch.org <mailto: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 <mailto: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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