-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
SIP wrote:
> Daniel,
Hi SIP.
> Check your stunaddr setting. Is it misspelled, or do they really use
> stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't believe
that this solves what I commented before about 192.168.1.2 direction, but,
just in case, I copy the output of debugging when trying to communicate to
ekiga.net.
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke
Max-Forwards: 70
To: <sip:8...@10.1.0.10>
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 709 INVITE
Contact: <sip:2...@10.1.0.65>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 933572867 1938524932 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
- --- (13 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request -
kafgeaflkmsd...@defiant.freesoftware.org
<--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
To: <sip:8...@10.1.0.10>;tag=as0a3a462b
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 709 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="497d879d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke
Max-Forwards: 70
To: <sip:8...@10.1.0.10>;tag=as0a3a462b
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 709 ACK
User-Agent: Twinkle/1.2
Content-Length: 0
<------------->
- --- (9 headers 0 lines) ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr
Max-Forwards: 70
Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="497d879d",uri="sip:8...@10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5
To: <sip:8...@10.1.0.10>
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 710 INVITE
Contact: <sip:2...@10.1.0.65>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 933572867 1938524932 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
- --- (14 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request -
kafgeaflkmsd...@defiant.freesoftware.org
Found user '201'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.0.65:8000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe (gsm|ulaw|
alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.0.65:8000
Looking for 8500 in from-internal (domain 10.1.0.10)
list_route: hop: <sip:2...@10.1.0.65>
<--- Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
To: <sip:8...@10.1.0.10>
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 710 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8...@10.1.0.10>
Content-Length: 0
<------------>
-- Executing [8...@from-internal:1] Dial("SIP/201-090ffff0",
"SIP/ekiga/500|20|r)") in new stack
Video is at 192.168.1.2 port 10112
Audio is at 192.168.1.2 port 12592
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40000 (h261) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.64.162.35:5060:
INVITE sip:5...@ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport
From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8
To: <sip:5...@ekiga.net>
Contact: <sip:2...@192.168.1.2>
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 21:30:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 331
v=0
o=root 4959 4959 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
b=CT:384
t=0 0
m=audio 12592 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10112 RTP/AVP 31
a=rtpmap:31 H261/90000
a=sendrecv
- ---
-- Called ekiga/500
<--- Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
To: <sip:8...@10.1.0.10>;tag=as37d19c71
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 710 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8...@10.1.0.10>
Content-Length: 0
<------------>
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK651d88ba;rport=10003;received=190.51.105.123
From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8
To: <sip:5...@ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="ekiga.net",
nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f"
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (9 headers 0 lines) ---
Transmitting (NAT) to 86.64.162.35:5060:
ACK sip:5...@ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport
From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8
To: <sip:5...@ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918
Contact: <sip:2...@192.168.1.2>
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
- ---
Video is at 192.168.1.2 port 10112
Audio is at 192.168.1.2 port 12592
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40000 (h261) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.64.162.35:5060:
INVITE sip:5...@ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport
From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8
To: <sip:5...@ekiga.net>
Contact: <sip:2...@192.168.1.2>
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="danib", realm="ekiga.net",
algorithm=MD5, uri="sip:5...@ekiga.net",
nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f",
response="152416b836f298095455859a7c3f1696"
Date: Mon, 17 Aug 2009 21:30:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 331
v=0
o=root 4959 4960 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
b=CT:384
t=0 0
m=audio 12592 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10112 RTP/AVP 31
a=rtpmap:31 H261/90000
a=sendrecv
- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport=10003;received=190.51.105.123
From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8
To: <sip:5...@ekiga.net>
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 103 INVITE
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;received=190.51.105.123;branch=z9hG4bK6e8ff8ba;rport=10003
Record-Route: <sip:86.64.162.35;lr=on>
From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8
To: <sip:5...@ekiga.net>;tag=as38bf28ad
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 103 INVITE
User-Agent: Ekiga.NET
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5...@86.64.162.35:5081>
Content-Type: application/sdp
Content-Length: 310
v=0
o=root 9963 9963 IN IP4 86.64.162.35
s=session
c=IN IP4 86.64.162.35
b=CT:384
t=0 0
m=audio 10400 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 14488 RTP/AVP 31
a=rtpmap:31 H261/90000
a=sendrecv
<------------->
- --- (13 headers 16 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found RTP video format 31
Peer audio RTP is at port 86.64.162.35:10400
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found video description format H261 for ID 31
Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0x40008 (alaw|
h261)/video=0x40000 (h261), combined - 0x40008 (alaw|h261)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 86.64.162.35:10400
Peer video RTP is at port 86.64.162.35:14488
list_route: hop: <sip:86.64.162.35;lr=on>
set_destination: Parsing <sip:86.64.162.35;lr=on> for address/port to send
to
set_destination: set destination to 86.64.162.35, port 5060
Transmitting (NAT) to 86.64.162.35:5060:
ACK sip:5...@86.64.162.35:5081 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK776caeb0;rport
Route: <sip:86.64.162.35;lr=on>
From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8
To: <sip:5...@ekiga.net>;tag=as38bf28ad
Contact: <sip:2...@192.168.1.2>
Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
- ---
-- SIP/ekiga-090cb900 answered SIP/201-090ffff0
Audio is at 10.1.0.10 port 12994
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
rom: "Hector" <sip:2...@10.1.0.10>;tag=typwm
To: <sip:8...@10.1.0.10>;tag=as37d19c71
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 710 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8...@10.1.0.10>
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 4959 4959 IN IP4 10.1.0.10
s=session
c=IN IP4 10.1.0.10
t=0 0
m=audio 12994 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKpnwhfosw
Max-Forwards: 70
Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="497d879d",uri="sip:8...@10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5
To: <sip:8...@10.1.0.10>;tag=as37d19c71
From: "Hector" <sip:2...@10.1.0.10>;tag=typwm
Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
CSeq: 710 ACK
User-Agent: Twinkle/1.2
Content-Length: 0
<------------->
- --- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:2...@10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK3c1a71ce;rport
From: "asterisk" <sip:aster...@10.1.0.10>;tag=as51f657b5
To: <sip:2...@10.1.0.65>
Contact: <sip:aster...@10.1.0.10>
Call-ID: 2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 21:30:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK3c1a71ce
To: <sip:2...@10.1.0.65>;tag=pecxh
From: "asterisk" <sip:aster...@10.1.0.10>;tag=as51f657b5
Call-ID: 2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0
<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10'
Method: OPTIONS
Reliably Transmitting (NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2268d402;rport
From: "asterisk" <sip:aster...@192.168.1.2>;tag=as2a2e4c13
To: <sip:ekiga.net>
Contact: <sip:aster...@192.168.1.2>
Call-ID: 1a9ce3ad2f18ecf129c457d527603...@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 21:30:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK2268d402;rport=10003;received=190.51.105.123
From: "asterisk" <sip:aster...@192.168.1.2:5060>;tag=as2a2e4c13
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c092
Call-ID: 1a9ce3ad2f18ecf129c457d527603...@192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '1a9ce3ad2f18ecf129c457d527603...@192.168.1.2'
Method: OPTIONS
Thanks for your reply.
Regards,
Daniel
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