[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 3 10:11:22 CST 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=5413
======================================================================
Reported By: mikma
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 5413
Category: Channels/chan_sip/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: confirmed
Target Version: 1.8
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 48491
Request Review:
======================================================================
Date Submitted: 2005-10-09 10:36 CDT
Last Modified: 2010-03-03 10:10 CST
======================================================================
Summary: [patch] [branch] Secure RTP (SRTP)
Description:
This patch adds initial support for secure RTP using libsrt[1]. It can
be used in for example an implementation of the sdecriptions draft[2].
[1] http://srtp.sourceforge.net/srtp.html
[2]
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions-12.txt
Update (17/12/2008): Branch against trunk is located here
http://svn.digium.com/svn/asterisk/team/group/srtp
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0010129 Module SRTP can't loaded
======================================================================
----------------------------------------------------------------------
(0118850) notthematrix (reporter) - 2010-03-03 10:10
https://issues.asterisk.org/view.php?id=5413#c118850
----------------------------------------------------------------------
@twilson
here the NEW output.
as you requested :)
<--- SIP read from TLS:92.254.55.200:2050 --->
INVITE sip:*66 at 87.251.43.124:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK818032521;rport;alias
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
To: <sip:*66 at 87.251.43.124:443;user=phone>
Call-ID: 237371426-51280-5 at 192.168.1.108
CSeq: 40 INVITE
Contact: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:31251788103 at 87.251.43.124:443;user=phone>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 631
v=0
o=31251788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 36774 RTP/SAVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:5Nx9/l0cXF/rQ9zCthJnod6m//F8r67o956ZvYct
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:A2wKgGpnnMbGiAqjSsC1smGUWGCF1BAzvAfSVcRZ
<------------->
--- (16 headers 22 lines) ---
Using INVITE request as basis request - 237371426-51280-5 at 192.168.1.108
Found peer '31251788103' for '31251788103' from 92.254.55.200:2050
<--- Reliably Transmitting (NAT) to 92.254.55.200:2050 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/TLS
192.168.1.108:51280;branch=z9hG4bK818032521;alias;received=92.254.55.200;rport=2050
f: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
t: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as25f1b441
i: 237371426-51280-5 at 192.168.1.108
CSeq: 40 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="3e7c7408"
l: 0
<------------>
Scheduling destruction of SIP dialog '237371426-51280-5 at 192.168.1.108' in
6400 ms (Method: INVITE)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
ACK sip:*66 at 87.251.43.124:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK818032521;rport;alias
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
To: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as25f1b441
Call-ID: 237371426-51280-5 at 192.168.1.108
CSeq: 40 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
INVITE sip:*66 at 87.251.43.124:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK1231987690;rport;alias
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
To: <sip:*66 at 87.251.43.124:443;user=phone>
Call-ID: 237371426-51280-5 at 192.168.1.108
CSeq: 41 INVITE
Contact: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
Authorization: Digest username="31251788103", realm="asterisk",
nonce="3e7c7408", uri="sip:*66 at 87.251.43.124:443;user=phone",
response="27167da0f1433490b60c72c9e6f04941", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:31251788103 at 87.251.43.124:443;user=phone>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 631
v=0
o=31251788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 36774 RTP/SAVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:5Nx9/l0cXF/rQ9zCthJnod6m//F8r67o956ZvYct
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:A2wKgGpnnMbGiAqjSsC1smGUWGCF1BAzvAfSVcRZ
<------------->
--- (17 headers 22 lines) ---
Sending to 92.254.55.200 : 2050 (NAT)
Using INVITE request as basis request - 237371426-51280-5 at 192.168.1.108
Found peer '31251788103' for '31251788103' from 92.254.55.200:2050
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G729E for ID 102
Found audio description format AAL2-G726-16 for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x2d0d
(g723|ulaw|alaw|g726|g729|ilbc|siren7)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0x10d (g723|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.108:36774
Looking for *66 in klant-31-123-123456 (domain 87.251.43.124)
list_route: hop:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
mastermetals1*CLI>
<--- Transmitting (NAT) to 92.254.55.200:2050 --->
SIP/2.0 100 Trying
v: SIP/2.0/TLS
192.168.1.108:51280;branch=z9hG4bK1231987690;alias;received=92.254.55.200;rport=2050
f: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
t: <sip:*66 at 87.251.43.124:443;user=phone>
i: 237371426-51280-5 at 192.168.1.108
CSeq: 41 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
m: <sip:*66 at 87.251.43.124:443;transport=TLS>
l: 0
<------------>
-- Executing [*66 at klant-31-123-123456:1]
Goto("SIP/31251788103-00000006", "from-internal,*66,1") in new stack
-- Goto (from-internal,*66,1)
-- Executing [*66 at from-internal:1] Answer("SIP/31251788103-00000006",
"") in new stack
Audio is at 87.251.43.124 port 17750
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
mastermetals1*CLI>
<--- Reliably Transmitting (NAT) to 92.254.55.200:2050 --->
SIP/2.0 200 OK
v: SIP/2.0/TLS
192.168.1.108:51280;branch=z9hG4bK1231987690;alias;received=92.254.55.200;rport=2050
f: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
t: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as0443c618
i: 237371426-51280-5 at 192.168.1.108
CSeq: 41 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
m: <sip:*66 at 87.251.43.124:443;transport=TLS>
c: application/sdp
l: 455
v=0
o=root 949742278 949742278 IN IP4 87.251.43.124
s=Asterisk PBX
c=IN IP4 87.251.43.124
t=0 0
m=audio 17750 RTP/SAVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:bW3qflnmvFJLgSW63DbwZX90CFmDyEpsd8I2LT7J
<------------>
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
ACK sip:*66 at 87.251.43.124:443;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK296795156;rport
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
To: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as0443c618
Call-ID: 237371426-51280-5 at 192.168.1.108
CSeq: 41 ACK
Contact: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
-- Executing [*66 at from-internal:2] Wait("SIP/31251788103-00000006",
"1") in new stack
-- Executing [*66 at from-internal:3] Macro("SIP/31251788103-00000006",
"user-callerid,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/31251788103-00000006",
"AMPUSER=31251788103") in new stack
-- Executing [s at macro-user-callerid:2]
GotoIf("SIP/31251788103-00000006", "0?report") in new stack
-- Executing [s at macro-user-callerid:3]
ExecIf("SIP/31251788103-00000006", "1?Set(REALCALLERIDNUM=31251788103)") in
new stack
-- Executing [s at macro-user-callerid:4] Set("SIP/31251788103-00000006",
"AMPUSER=") in new stack
-- Executing [s at macro-user-callerid:5] Set("SIP/31251788103-00000006",
"AMPUSERCIDNAME=") in new stack
-- Executing [s at macro-user-callerid:6]
GotoIf("SIP/31251788103-00000006", "1?report") in new stack
-- Goto (macro-user-callerid,s,9)
-- Executing [s at macro-user-callerid:9]
GotoIf("SIP/31251788103-00000006", "0?continue") in new stack
-- Executing [s at macro-user-callerid:10]
Set("SIP/31251788103-00000006", "__TTL=64") in new stack
-- Executing [s at macro-user-callerid:11]
GotoIf("SIP/31251788103-00000006", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s at macro-user-callerid:18]
NoOp("SIP/31251788103-00000006", "Using CallerID "device" <31251788103>")
in new stack
-- Executing [*66 at from-internal:4] DBdel("SIP/31251788103-00000006",
"CP/") in new stack
-- DBdel: family=CP, key=
-- DBdel: Error deleting key from database.
-- Executing [*66 at from-internal:5]
Playback("SIP/31251788103-00000006", "privacy-your-callerid-is&activated")
in new stack
-- <SIP/31251788103-00000006> Playing 'privacy-your-callerid-is.alaw'
(language 'en')
-- <SIP/31251788103-00000006> Playing 'activated.alaw' (language
'en')
-- Executing [*66 at from-internal:6] Macro("SIP/31251788103-00000006",
"hangupcall,") in new stack
-- Executing [s at macro-hangupcall:1]
ResetCDR("SIP/31251788103-00000006", "w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/31251788103-00000006",
"") in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/31251788103-00000006",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/31251788103-00000006",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/31251788103-00000006",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11]
Hangup("SIP/31251788103-00000006", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/31251788103-00000006' in macro 'hangupcall'
== Spawn extension (from-internal, *66, 6) exited non-zero on
'SIP/31251788103-00000006'
-- Executing [h at from-internal:1] Macro("SIP/31251788103-00000006",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1]
ResetCDR("SIP/31251788103-00000006", "w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/31251788103-00000006",
"") in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/31251788103-00000006",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/31251788103-00000006",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/31251788103-00000006",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11]
Hangup("SIP/31251788103-00000006", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/31251788103-00000006' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/31251788103-00000006'
Scheduling destruction of SIP dialog '237371426-51280-5 at 192.168.1.108' in
6400 ms (Method: INVITE)
set_destination: Parsing
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone> for
address/port to send to
set_destination: set destination to 192.168.1.108, port 51280
Reliably Transmitting (NAT) to 92.254.55.200:2050:
BYE sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone SIP/2.0
v: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK6f54fabc;rport
Max-Forwards: 70
f: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as0443c618
t: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
i: 237371426-51280-5 at 192.168.1.108
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="31251788103", realm="asterisk",
algorithm=MD5, uri="87.251.43.124", nonce="",
response="36cb4d85b80aec9a89c3e3eb1736be73"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
l: 0
---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK6f54fabc;rport=443
From: <sip:*66 at 87.251.43.124:443;user=phone>;tag=as0443c618
To: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1150824446
Call-ID: 237371426-51280-5 at 192.168.1.108
CSeq: 102 BYE
Contact: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
Supported: replaces, path, timer
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '237371426-51280-5 at 192.168.1.108' Method:
INVITE
Reliably Transmitting (NAT) to 92.254.55.200:2050:
OPTIONS sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone
SIP/2.0
v: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK2efeb8f9;rport
Max-Forwards: 70
f: "Unknown" <sip:Unknown at 87.251.43.124:443>;tag=as5d582010
t: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
m: <sip:Unknown at 87.251.43.124:443;transport=TLS>
i: 3547b99d7a4becd56c3d6c0341b73a25 at 87.251.43.124
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 03 Mar 2010 16:08:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0
---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK2efeb8f9;rport=443
From: "Unknown" <sip:Unknown at 87.251.43.124:443>;tag=as5d582010
To:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>;tag=1598626929
Call-ID: 3547b99d7a4becd56c3d6c0341b73a25 at 87.251.43.124
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
REGISTER sip:87.251.43.124:443 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK11023348;rport;alias
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1970777137
To: <sip:31251788103 at 87.251.43.124:443;user=phone>
Call-ID: 1562417533-51280-1 at 192.168.1.108
CSeq: 2030 REGISTER
Contact:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B82131B12>"
Authorization: Digest username="31251788103", realm="asterisk",
nonce="2c37f64b", uri="sip:87.251.43.124:443",
response="dd43fdc1b25dadc7b82d0e525c6eb471", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Supported: path
Expires: 120
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
mastermetals1*CLI>
<--- Transmitting (NAT) to 92.254.55.200:2050 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/TLS
192.168.1.108:51280;branch=z9hG4bK11023348;alias;received=92.254.55.200;rport=2050
f: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1970777137
t: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=as5eb1410a
i: 1562417533-51280-1 at 192.168.1.108
CSeq: 2030 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="234309ec"
l: 0
<------------>
Scheduling destruction of SIP dialog '1562417533-51280-1 at 192.168.1.108' in
32000 ms (Method: REGISTER)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
REGISTER sip:87.251.43.124:443 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:51280;branch=z9hG4bK109093412;rport;alias
From: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1970777137
To: <sip:31251788103 at 87.251.43.124:443;user=phone>
Call-ID: 1562417533-51280-1 at 192.168.1.108
CSeq: 2031 REGISTER
Contact:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B82131B12>"
Authorization: Digest username="31251788103", realm="asterisk",
nonce="234309ec", uri="sip:87.251.43.124:443",
response="ea7907d8e3b45eb63708ab53998f366f", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Supported: path
Expires: 120
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 92.254.55.200 : 2050 (NAT)
Reliably Transmitting (NAT) to 92.254.55.200:2050:
OPTIONS sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone
SIP/2.0
v: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK6f033c81;rport
Max-Forwards: 70
f: "Unknown" <sip:Unknown at 87.251.43.124:443>;tag=as47a0d79e
t: <sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>
m: <sip:Unknown at 87.251.43.124:443;transport=TLS>
i: 014717b27c4e9f2f0fed05ba3f13ea3b at 87.251.43.124
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 03 Mar 2010 16:08:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0
---
mastermetals1*CLI>
<--- Transmitting (NAT) to 92.254.55.200:2050 --->
SIP/2.0 200 OK
v: SIP/2.0/TLS
192.168.1.108:51280;branch=z9hG4bK109093412;alias;received=92.254.55.200;rport=2050
f: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=1970777137
t: <sip:31251788103 at 87.251.43.124:443;user=phone>;tag=as5eb1410a
i: 1562417533-51280-1 at 192.168.1.108
CSeq: 2031 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
Expires: 120
m:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>;expires=120
Date: Wed, 03 Mar 2010 16:08:04 GMT
l: 0
<------------>
Scheduling destruction of SIP dialog '1562417533-51280-1 at 192.168.1.108' in
32000 ms (Method: REGISTER)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.55.200:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.251.43.124:443;branch=z9hG4bK6f033c81;rport=443
From: "Unknown" <sip:Unknown at 87.251.43.124:443>;tag=as47a0d79e
To:
<sip:31251788103 at 192.168.1.108:51280;transport=tls;user=phone>;tag=488688169
Call-ID: 014717b27c4e9f2f0fed05ba3f13ea3b at 87.251.43.124
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE
Content-Length: 0
Issue History
Date Modified Username Field Change
======================================================================
2010-03-03 10:10 notthematrix Note Added: 0118850
======================================================================
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