Here is the pjsipua output, it tells nat type is symmetric and
manage to call through with stun (which is not supposed
to work with symmetric nats?) :
20:23:52.354 os_core_unix.c pjlib 0.9.0-trunk for POSIX initialized
20:24:02.363 sip_endpoint.c Creating endpoint instance...
20:24:02.367 pjlib select() I/O Queue created (0x8190eb8)
20:24:02.367 sip_endpoint.c Module "mod-msg-print" registered
20:24:02.367 sip_transport. Transport manager created.
20:24:02.396 sip_endpoint.c Module "mod-pjsua-log" registered
20:24:02.396 sip_endpoint.c Module "mod-tsx-layer" registered
20:24:02.396 sip_endpoint.c Module "mod-stateful-util" registered
20:24:02.396 sip_endpoint.c Module "mod-ua" registered
20:24:02.397 sip_endpoint.c Module "mod-100rel" registered
20:24:02.397 sip_endpoint.c Module "mod-pjsua" registered
20:24:02.397 sip_endpoint.c Module "mod-invite" registered
20:24:02.397 pjsua_core.c STUN server 88.88.88.89 resolved, address
is 88.88.88.89:3478
20:24:02.419 pasound.c PortAudio sound library initialized,
status=0
20:24:02.419 pasound.c PortAudio host api count=1
20:24:02.419 pasound.c Sound device count=1
20:24:02.419 pjlib select() I/O Queue created (0x8197d84)
20:24:02.433 sip_endpoint.c Module "mod-evsub" registered
20:24:02.433 sip_endpoint.c Module "mod-presence" registered
20:24:02.433 sip_endpoint.c Module "mod-refer" registered
20:24:02.434 sip_endpoint.c Module "mod-pjsua-pres" registered
20:24:02.434 sip_endpoint.c Module "mod-pjsua-im" registered
20:24:02.434 sip_endpoint.c Module "mod-pjsua-options" registered
20:24:02.434 pjsua_core.c 1 SIP worker threads created
20:24:02.434 pjsua_core.c pjsua version 0.9.0-trunk for
i686-pc-linux-gnu initialized
20:24:02.537 pjsua_core.c SIP UDP socket reachable at
88.88.88.88:5060
20:24:02.537 udp0x81a9df8 SIP UDP transport started, published
address is 88.88.88.88:5060
20:24:02.537 pjsua_acc.c Account <sip:88.88.88.88:5060> added with
id 0
20:24:10.383 stuntsx0x81aa9 STUN timeout waiting for response
20:24:10.439 stuntsx0x81ab1 STUN timeout waiting for response
20:24:10.484 pjsua_app.c NAT detected as Symmetric
20:24:12.535 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.1.11:5060
20:24:12.535 pjsua_acc.c Account
<sip:192.168.1.11:5060;transport=TCP> added with id 1
20:24:12.535 pjsua_acc.c Account sip:pjsipua at 88.88.88.91 added with
id 2
20:24:12.536 pjsua_core.c TX 435 bytes Request msg
REGISTER/cseq=36785 (tdta0x81a5818) to UDP 88.88.88.91:5060:
REGISTER sip:88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjSqYr27J4PZxlFQvoP4mYiEwLBslHU4Tp
Max-Forwards: 70
From: <sip:pjsipua@88.88.88.91>;tag=nytk3c1Tl4nQ87KqEk8u9Ef7cCcQOXWs
To: <sip:pjsipua at 88.88.88.91>
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36785 REGISTER
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Contact: <sip:pjsipua at 88.88.88.88:5060>
Expires: 300
Content-Length: 0
--end msg--
20:24:12.536 pjsua_acc.c Registration sent
20:24:12.619 pjsua_media.c RTP socket reachable at 88.88.88.88:4000
20:24:12.619 pjsua_media.c RTCP socket reachable at 88.88.88.88:4001
20:24:12.740 pjsua_core.c RX 554 bytes Response msg
401/REGISTER/cseq=36785 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjSqYr27J4PZxlFQvoP4mYiEwLBslHU4Tp;received=88.88.88.88;rport=5060
From: <sip:pjsipua@88.88.88.91>;tag=nytk3c1Tl4nQ87KqEk8u9Ef7cCcQOXWs
To: <sip:pjsipua at 88.88.88.91>;tag=as67921369
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36785 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="4edab1e2"
Content-Length: 0
--end msg--
20:24:12.740 pjsua_core.c TX 596 bytes Request msg
REGISTER/cseq=36786 (tdta0x81a5818) to UDP 88.88.88.91:5060:
REGISTER sip:88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjIZYldNSI4EHx4memOMrjadGcp8bhVeD6
Max-Forwards: 70
From: <sip:pjsipua@88.88.88.91>;tag=nytk3c1Tl4nQ87KqEk8u9Ef7cCcQOXWs
To: <sip:pjsipua at 88.88.88.91>
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36786 REGISTER
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Contact: <sip:pjsipua at 88.88.88.88:5060>
Expires: 300
Authorization: Digest username="pjsipua", realm="asterisk",
nonce="4edab1e2", uri="sip:88.88.88.91",
response="acf322d359b767ed02194c4f2a7d50ff", algorithm=MD5
Content-Length: 0
--end msg--
20:24:12.748 pjsua_media.c RTP socket reachable at 88.88.88.88:4002
20:24:12.748 pjsua_media.c RTCP socket reachable at 88.88.88.88:4003
20:24:12.862 pjsua_core.c RX 572 bytes Response msg
200/REGISTER/cseq=36786 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjIZYldNSI4EHx4memOMrjadGcp8bhVeD6;received=88.88.88.88;rport=5060
From: <sip:pjsipua@88.88.88.91>;tag=nytk3c1Tl4nQ87KqEk8u9Ef7cCcQOXWs
To: <sip:pjsipua at 88.88.88.91>;tag=as67921369
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36786 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 300
Contact: <sip:pjsipua at 88.88.88.88:5060>;expires=300
Date: Mon, 04 Aug 2008 18:23:21 GMT
Content-Length: 0
--end msg--
20:24:12.899 pjsua_media.c RTP socket reachable at 88.88.88.88:4004
20:24:12.899 pjsua_media.c RTCP socket reachable at 88.88.88.88:4005
20:24:13.059 pjsua_media.c RTP socket reachable at 88.88.88.88:4006
20:24:13.059 pjsua_media.c RTCP socket reachable at 88.88.88.88:4007
>>>>
Account list:
[ 0] <sip:88.88.88.88:5060>: does not register
Online status: Online
[ 1] <sip:192.168.1.11:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:pjsipua at 88.88.88.91: 100/In Progress (expires=294)
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account:
|
| | |
|
| m Make new call | +b Add new buddy .| +a Add new
accnt |
| M Make multiple calls | -b Delete buddy | -a Delete
accnt. |
| a Answer call | i Send IM | !a Modify
accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr
(Re-)register |
| H Hold call | u Unsubscribe presence | ru
Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle
next ac.|
| U send UPDATE | T Set online status | < Cycle
prev ac.|
| ],[ Select next/prev call
+--------------------------+-------------------+
| x Xfer call | Media Commands: | Status &
Config: |
| X Xfer with Replaces | |
|
| # Send RFC 2833 DTMF | cl List ports | d Dump
status |
| * Send DTMF with INFO | cc Connect port | dd Dump
detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump
config |
| | V Adjust audio Volume | f Save
config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save
config |
+------------------------------+--------------------------+-------------------+
| q QUIT sleep MS echo [0|1|txt] n: detect NAT type
|
+=============================================================================+
You have 0 active call
>>> 20:24:13.060 pjsua_acc.c sip:pjsipua at 88.88.88.91: registration
success, status=200 (OK), will re-register in 300 seconds
20:24:13.060 pjsua_acc.c Keep-alive timer started for acc 2,
destination:88.88.88.91:5060, interval:15s
(You currently have 0 calls)
Buddy list:
-none-
Choices:
0 For current dialog.
-1 All 0 buddies in buddy list
[1 - 0] Select from buddy list
URL An URL
<Enter> Empty input (or 'q') to cancel
Make call: 20:24:38.866 pjsua_media.c pjsua_set_snd_dev(): attempting
to open devices @16000 Hz
20:24:38.912 echo_speex.c Speex Echo canceller/AEC created,
clock_rate=16000, samples per frame=320, tail length=200 ms, latency=192
ms
20:24:38.912 pjsua_call.c Making call with acc #2 to
sip:3 at 88.88.88.91
20:24:38.947 pjsua_media.c Media index 0 selected for call 0
20:24:38.948 pjsua_core.c TX 1032 bytes Request msg
INVITE/cseq=27751 (tdta0x81e69c0) to UDP 88.88.88.91:5060:
INVITE sip:3 at 88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjvbVltz-ZS2a3-5ZTu33eAzKhvunZqkmX
Max-Forwards: 70
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91
Contact: <sip:pjsipua at 88.88.88.88:5060>
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27751 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3426863078 3426863078 IN IP4 88.88.88.88
s=pjmedia
c=IN IP4 88.88.88.88
t=0 0
a=X-nat:6
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101
a=rtcp:4001 IN IP4 88.88.88.88
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--end msg--
20:24:38.949 pjsua_app.c Call 0 state changed to CALLING
>>> 20:24:39.068 pjsua_core.c RX 542 bytes Response msg
401/INVITE/cseq=27751 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjvbVltz-ZS2a3-5ZTu33eAzKhvunZqkmX;received=88.88.88.88;rport=5060
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as5fb8e742
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27751 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="76f7b672"
Content-Length: 0
--end msg--
20:24:39.068 pjsua_core.c TX 327 bytes Request msg ACK/cseq=27751
(tdta0x81e9178) to UDP 88.88.88.91:5060:
ACK sip:3 at 88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjvbVltz-ZS2a3-5ZTu33eAzKhvunZqkmX
Max-Forwards: 70
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as5fb8e742
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27751 ACK
Content-Length: 0
--end msg--
20:24:39.069 pjsua_core.c TX 1195 bytes Request msg
INVITE/cseq=27752 (tdta0x81e69c0) to UDP 88.88.88.91:5060:
INVITE sip:3 at 88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjDSaYElBAXozKkr5vny9DlghfEEGEHU7l
Max-Forwards: 70
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91
Contact: <sip:pjsipua at 88.88.88.88:5060>
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27752 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Authorization: Digest username="pjsipua", realm="asterisk",
nonce="76f7b672", uri="sip:3 at 88.88.88.91",
response="38af789bddd147cb2f2c32e8fd430cbb", algorithm=MD5
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3426863078 3426863078 IN IP4 88.88.88.88
s=pjmedia
c=IN IP4 88.88.88.88
t=0 0
a=X-nat:6
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101
a=rtcp:4001 IN IP4 88.88.88.88
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--end msg--
20:24:39.069 pjsua_app.c Call 0 state changed to CALLING
20:24:39.227 pjsua_core.c RX 475 bytes Response msg
100/INVITE/cseq=27752 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjDSaYElBAXozKkr5vny9DlghfEEGEHU7l;received=88.88.88.88;rport=5060
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27752 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:3 at 88.88.88.91>
Content-Length: 0
--end msg--
20:24:39.228 pjsua_core.c RX 784 bytes Response msg
200/INVITE/cseq=27752 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjDSaYElBAXozKkr5vny9DlghfEEGEHU7l;received=88.88.88.88;rport=5060
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as40556411
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27752 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:3 at 88.88.88.91>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2081192799 2081192799 IN IP4 88.88.88.91
s=Asterisk PBX 1.6.0-beta9
c=IN IP4 88.88.88.91
t=0 0
m=audio 11068 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--end msg--
20:24:39.228 pjsua_app.c Call 0 state changed to CONNECTING
20:24:39.228 strm0x81ec5ec VAD temporarily disabled
20:24:39.228 strm0x81ec5ec Encoder stream started
20:24:39.228 strm0x81ec5ec Decoder stream started
20:24:39.228 pjsua_media.c Media updates, stream #0: PCMU (sendrecv)
20:24:39.228 conference.c Port 3 (sip:3 at 88.88.88.91) transmitting to
port 0 (/dev/dsp)
20:24:39.228 conference.c Port 0 (/dev/dsp) transmitting to port 3
(sip:3 at 88.88.88.91)
20:24:39.228 pjsua_app.c Media for call 0 is active
20:24:39.228 pjsua_core.c TX 327 bytes Request msg ACK/cseq=27752
(tdta0x81f0970) to UDP 88.88.88.91:5060:
ACK sip:3 at 88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjzdjOv0aIgctKp8I2FiuHYHRs5rEr7mBb
Max-Forwards: 70
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as40556411
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27752 ACK
Content-Length: 0
--end msg--
20:24:39.228 pjsua_app.c Call 0 state changed to CONFIRMED
20:24:39.229 Master/sound Underflow, buf_cnt=0, will generate 1
frame
20:24:39.837 strm0x81ec5ec VAD re-enabled
20:25:11.924 pjsua_core.c TX 377 bytes Request msg BYE/cseq=27753
(tdta0x81e69c0) to UDP 88.88.88.91:5060:
BYE sip:3 at 88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjllDeEPJGO5hAG-sYL0JmgaKo93pw.65E
Max-Forwards: 70
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as40556411
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27753 BYE
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Content-Length: 0
--end msg--
>>> 20:25:12.003 pjsua_core.c RX 483 bytes Response msg
200/BYE/cseq=27753 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjllDeEPJGO5hAG-sYL0JmgaKo93pw.65E;received=88.88.88.88;rport=5060
From: sip:pjsipua@88.88.88.91;tag=sxl.nJYo0hhLRcDCCUlp2aQTj8eSIblI
To: sip:3 at 88.88.88.91;tag=as40556411
Call-ID: KSU-faAAZcrhUE9oLq5HBkROb5YiIKR4
CSeq: 27753 BYE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:3 at 88.88.88.91>
Content-Length: 0
--end msg--
20:25:12.003 pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal
call clearing)]
20:25:12.004 pjsua_app.c
[DISCONNCTD] To: sip:3 at 88.88.88.91;tag=as40556411
Call time: 00h:00m:32s, 1st res in 316 ms, conn in 316ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMU @8KHz, sendrecv, peer=88.88.88.91:11068
RX pt=0, stat last update: 00h:00m:00.938s ago
total 1.6Kpkt 258.2KB (322.8KB +IP hdr) @avg=63.0Kbps/78.7Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 1.254 39.875 17.500 3.833
TX pt=0, ptime=20ms, stat last update: 00h:00m:02.690s ago
total 94pkt 15.0KB (18.8KB +IP hdr) @avg 3.6Kbps/4.5Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 2.625 21.291 61.875 2.625 20.640
RTT msec : 39.825 67.567 160.383 44.815 65.286
20:25:12.004 pjsua_media.c Media session for call 0 is destroyed
20:25:17.004 sound_port.c EC suspended because of inactivity
20:25:17.459 pjsua_core.c TX 433 bytes Request msg
REGISTER/cseq=36787 (tdta0x81a7aa8) to UDP 88.88.88.91:5060:
REGISTER sip:88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjdu4QOpP-jqgMa-ybRRqn3dgnPQ9Mj2Ju
Max-Forwards: 70
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36787 REGISTER
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Contact: <sip:pjsipua at 88.88.88.88:5060>
Expires: 0
Content-Length: 0
--end msg--
20:25:17.460 pjsua_acc.c Unregistration sent
20:25:17.460 pjsua_media.c Closing (null) sound playback device and
(null) sound capture device
20:25:18.963 pasound.c PortAudio sound library shutting down..
20:25:18.963 pjsua_core.c Shutting down...
20:25:18.963 pjsua_core.c TX 433 bytes Request msg
REGISTER/cseq=36787 (tdta0x81a7aa8) to UDP 88.88.88.91:5060:
REGISTER sip:88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjdu4QOpP-jqgMa-ybRRqn3dgnPQ9Mj2Ju
Max-Forwards: 70
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36787 REGISTER
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Contact: <sip:pjsipua at 88.88.88.88:5060>
Expires: 0
Content-Length: 0
--end msg--
20:25:18.963 pjsua_core.c RX 554 bytes Response msg
401/REGISTER/cseq=36787 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjdu4QOpP-jqgMa-ybRRqn3dgnPQ9Mj2Ju;received=88.88.88.88;rport=5060
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>;tag=as619787bc
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36787 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="64e8c7ca"
Content-Length: 0
--end msg--
20:25:18.963 pjsua_core.c TX 594 bytes Request msg
REGISTER/cseq=36788 (tdta0x81a7aa8) to UDP 88.88.88.91:5060:
REGISTER sip:88.88.88.91 SIP/2.0
Via: SIP/2.0/UDP
88.88.88.88:5060;rport;branch=z9hG4bKPjHeWs2he9sjzHZHLnn8RgOhEHQKcrfzzD
Max-Forwards: 70
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36788 REGISTER
User-Agent: PJSUA v0.9.0-trunk/i686-pc-linux-gnu
Contact: <sip:pjsipua at 88.88.88.88:5060>
Expires: 0
Authorization: Digest username="pjsipua", realm="asterisk",
nonce="64e8c7ca", uri="sip:88.88.88.91",
response="b12706028ea0534e1d701e9c212f2349", algorithm=MD5
Content-Length: 0
--end msg--
20:25:19.082 pjsua_core.c RX 554 bytes Response msg
401/REGISTER/cseq=36787 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjdu4QOpP-jqgMa-ybRRqn3dgnPQ9Mj2Ju;received=88.88.88.88;rport=5060
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>;tag=as619787bc
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36787 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="64e8c7ca"
Content-Length: 0
--end msg--
20:25:19.082 pjsua_core.c RX 517 bytes Response msg
200/REGISTER/cseq=36788 (rdata0x81ab474) from UDP 88.88.88.91:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
88.88.88.88:5060;branch=z9hG4bKPjHeWs2he9sjzHZHLnn8RgOhEHQKcrfzzD;received=88.88.88.88;rport=5060
From: <sip:pjsipua@88.88.88.91>;tag=v7dSIrnoxWWsy0Qdu7OStz6vSDZ9Of0M
To: <sip:pjsipua at 88.88.88.91>;tag=as619787bc
Call-ID: euPkL.fUaHRHRDwhxOAptnbdNRBJqU66
CSeq: 36788 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 0
Date: Mon, 04 Aug 2008 18:24:27 GMT
Content-Length: 0
--end msg--
20:25:19.082 pjsua_acc.c sip:pjsipua at 88.88.88.91: unregistration
success
20:25:19.967 pjsua_core.c Destroying...
20:25:19.967 sip_transactio Stopping transaction layer module
20:25:19.968 sip_endpoint.c Module "mod-pjsua-options" unregistered
20:25:19.968 sip_endpoint.c Module "mod-pjsua-im" unregistered
20:25:19.968 sip_endpoint.c Module "mod-pjsua-pres" unregistered
20:25:19.968 sip_endpoint.c Module "mod-pjsua" unregistered
20:25:19.968 sip_endpoint.c Module "mod-stateful-util" unregistered
20:25:19.968 sip_endpoint.c Module "mod-refer" unregistered
20:25:19.968 sip_endpoint.c Module "mod-presence" unregistered
20:25:19.968 sip_endpoint.c Module "mod-evsub" unregistered
20:25:19.968 sip_endpoint.c Module "mod-invite" unregistered
20:25:19.968 sip_endpoint.c Module "mod-100rel" unregistered
20:25:19.968 sip_endpoint.c Module "mod-ua" unregistered
20:25:19.968 sip_transactio Transaction layer module destroyed
20:25:19.968 sip_endpoint.c Module "mod-tsx-layer" unregistered
20:25:19.968 sip_endpoint.c Module "mod-msg-print" unregistered
20:25:19.968 sip_endpoint.c Module "mod-pjsua-log" unregistered
20:25:19.971 tcplis:5060 SIP TCP listener destroyed
20:25:19.971 sip_endpoint.c Endpoint 0x818900c destroyed
20:25:19.971 pjsua_core.c PJSUA destroyed...
Can you tell me if am I wrong ?
Thanks
Le lundi 04 ao?t 2008 ? 14:23 +0100, Benny Prijono a ?crit :
> I think I'd have to see the pjsua log (at level 5) to know what's
> going on. And I assume you're using the latest (0.9) pjsip, since I
> recall that we've had wrong detection algorithm in the rather old
> pjsip version.
>
> Btw lets use the mailing list for discussions like this so that it
> would benefit others.
>
> Cheers
> Benny
>
> On Mon, Aug 4, 2008 at 12:59 PM, Philippe HENSEL
> <philippe.hensel at uha.fr> wrote:
> Hi Benny,
>
> I've created a small STUN lab with 4 Virtual Machines :
> - One with PjSipUa
> - One as Linux NAT (using MASQUERADE)
> - One with asterisk
> - One with STUN Server (Vovida implementation)
>
> PjSipUa detects Linux NAT as "Symmetric NAT" where
> jstun client detects "Port Restricted NAT" ...
>
> When doing a netstat-nat on the NAT Virtual Machine I
> can see that MASQUERADE allocate same public address
> for 2 differentes Destination Addresses ...
> So I think that this is not "Symmetric NAT"...
>
> Can you tell me if am I wrong or right ?
>
> Thanks
>
>
>
>
