Re: [Freeswitch-dev] Call to registered phone behind NAT

23 May 2010

      Brian West wrote:
...
I really find that one hard to believe.
Hard to believe, but this is true:

I have made PURE experiment.
My friend has registered SIP-phone behind NAT in Ukraine.
His registration:

Call-ID:        4a978b30@192.168.10.103
User:           1012@192.168.1.5
Contact:        "user" <sip:1012@192.168.1.136:1024>
Agent:          AcctonVoIP/2.5
Status:         Registered(UDP)(unknown) EXP(2010-05-23 14:27:20)
Host:           olegh.ath.cx
IP:             91.207.244.1
Port:           1024
Auth-User:      1012
Auth-Realm:     olegh.ath.cx
MWI-Account:    1012@192.168.1.5

I tried call 1001 -> 1012, fs_cli prints:

2010-05-23 12:37:01.593320 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/sip:1012@192.168.1.136:1024 
[2b79d969-8966-df11-ac63-005004c3cb7e]

I sniffed network traffic, FS tries connect to his LAN address from 
"Contact", not his real address:

olegh# tcpdump host'(192.168.1.136 or 91.207.244.1)'
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on xl0, link-type EN10MB (Ethernet), capture size 96 bytes
12:40:55.044208 arp who-has 192.168.1.136 tell deskpro.khovayko.com
12:40:55.544711 arp who-has 192.168.1.136 tell deskpro.khovayko.com
12:40:56.545145 arp who-has 192.168.1.136 tell deskpro.khovayko.com
12:40:58.546827 arp who-has 192.168.1.136 tell deskpro.khovayko.com
...
sofia profile xxx siptrace on
I will be you its NOT ignoring the fs_path you just can't see the 
packet leaving and since its TCP the far side nat I suspect has closed 
the nat translation thus you'll need to register more often or enable 
keep alives on the device.
/b
I tried to do this -- results as same as with tcpdump.
My local phone has number 1001, IP=192.168.1.130, try to call 1012 
(behind NAT, see registration above):

freeswitch@internal>
freeswitch@internal>
freeswitch@internal>
recv 950 bytes from udp/[192.168.1.130]:5060 at 16:55:30.811681:
    ------------------------------------------------------------------------
    INVITE sip:1012@192.168.1.5 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bKe2ba0d5f00302
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012<sip:1012@192.168.1.5>
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166667 INVITE
    Max-Forwards: 70
    Contact: 1001 Cabinet<sip:1001@192.168.1.130:5060>
    P-Preferred-Identity: 1001 Cabinet<sip:1001@192.168.1.5:5060>
    User-Agent: AcctonVoIP/2.5
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, 
PRACK, UPDATE, MESSAGE, SERVICE, INFO, PING
    Allow-Events: refer
    Supported: 100rel, replaces
    Content-Length: 322
    Content-Type: application/sdp

    v=0
    o=- 608413784 756898537 IN IP4 192.168.1.130
    s=Voice call
    e=you@yourcompany.com
    c=IN IP4 192.168.1.130
    t=0 0
    a=sendrecv
    m=audio 5000 RTP/AVP 0 8 18 100
    a=ptime:20
    a=maxptime:30
    a=sendrecv
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:18 g729/8000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15
    ------------------------------------------------------------------------
send 307 bytes to udp/[192.168.1.130]:5060 at 16:55:30.815843:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bKe2ba0d5f00302
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012<sip:1012@192.168.1.5>
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166667 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
    Content-Length: 0

    ------------------------------------------------------------------------
send 788 bytes to udp/[192.168.1.130]:5060 at 16:55:30.842061:
    ------------------------------------------------------------------------
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bKe2ba0d5f00302
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012 <sip:1012@192.168.1.5>;tag=Za0U6NHjK7H6S
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166667 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, presence, dialog, line-seize, call-info, sla, 
include-session-description, presence.winfo, message-summary, refer
    Proxy-Authenticate: Digest realm="192.168.1.5", 
nonce="4f2df2fe-8b66-df11-ac63-005004c3cb7e", algorithm=MD5, qop="auth"
    Content-Length: 0

    ------------------------------------------------------------------------
recv 329 bytes from udp/[192.168.1.130]:5060 at 16:55:30.907144:
    ------------------------------------------------------------------------
    ACK sip:1012@192.168.1.5 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bKe2ba0d5f00302
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012<sip:1012@192.168.1.5>;tag=Za0U6NHjK7H6S
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166667 ACK
    Max-Forwards: 70
    User-Agent: AcctonVoIP/2.5
    Content-Length: 0

    ------------------------------------------------------------------------
recv 1177 bytes from udp/[192.168.1.130]:5060 at 16:55:30.913127:
    ------------------------------------------------------------------------
    INVITE sip:1012@192.168.1.5 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK69429b9100303
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012<sip:1012@192.168.1.5>
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166668 INVITE
    Max-Forwards: 70
    Contact: 1001 Cabinet<sip:1001@192.168.1.130:5060>
    P-Preferred-Identity: 1001 Cabinet<sip:1001@192.168.1.5:5060>
    User-Agent: AcctonVoIP/2.5
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, 
PRACK, UPDATE, MESSAGE, SERVICE, INFO, PING
    Allow-Events: refer
    Supported: 100rel, replaces
    Proxy-Authorization: Digest username="1001", realm="192.168.1.5", 
uri="sip:1012@192.168.1.5", 
nonce="4f2df2fe-8b66-df11-ac63-005004c3cb7e", qop=auth, 
cnonce="2f9b946d", nc=00000001, response="32cf7ea1d398bce1e1c8f53380a6bcea"
    Content-Length: 322
    Content-Type: application/sdp

    v=0
    o=- 608413784 756898537 IN IP4 192.168.1.130
    s=Voice call
    e=you@yourcompany.com
    c=IN IP4 192.168.1.130
    t=0 0
    a=sendrecv
    m=audio 5000 RTP/AVP 0 8 18 100
    a=ptime:20
    a=maxptime:30
    a=sendrecv
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:18 g729/8000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15
    ------------------------------------------------------------------------
send 307 bytes to udp/[192.168.1.130]:5060 at 16:55:30.919399:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK69429b9100303
    From: 1001 Cabinet<sip:1001@192.168.1.5>;tag=4df567c4
    To: 1012<sip:1012@192.168.1.5>
    Call-ID: 78fdee2b-17@192.168.1.130
    CSeq: 1380166668 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
    Content-Length: 0

    ------------------------------------------------------------------------
send 1302 bytes to udp/[192.168.1.136]:1024 at 16:55:31.703579:
    ------------------------------------------------------------------------
    INVITE sip:1012@192.168.1.136:1024 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.5;rport;branch=z9hG4bK2ajD0jFj617vN
    Max-Forwards: 69
    From: "Extension 1001" <sip:1001@192.168.1.5>;tag=1vjDacKSDSyBH
    To: <sip:1012@192.168.1.136:1024>
    Call-ID: d6d76003-e12e-122d-728c-2fb53066047d
    CSeq: 131198529 INVITE
    Contact: <sip:mod_sofia@192.168.1.5:5060>
    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, presence, dialog, line-seize, call-info, sla, 
include-session-description, presence.winfo, message-summary, refer
    Privacy: none
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 425
    X-FS-Support: update_display
    P-Asserted-Identity: "Extension 1001" <sip:1001@192.168.1.5>

    v=0
    o=FreeSWITCH 1274609641 1274609642 IN IP4 192.168.1.5
    s=FreeSWITCH
    c=IN IP4 192.168.1.5
    t=0 0
    m=audio 24090 RTP/AVP 0 115 107 9 8 3 101 13
    a=rtpmap:0 PCMU/8000
    a=rtpmap:115 G7221/32000
    a=fmtp:115 bitrate=48000
    a=rtpmap:107 G7221/16000
    a=fmtp:107 bitrate=32000
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=ptime:20
    ------------------------------------------------------------------------

Re: [Freeswitch-dev] Call to registered phone behind NAT

Oleg Khovayko