[asterisk-bugs] [Asterisk 0005413]: [patch] [branch] Secure RTP (SRTP)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 3 09:25:53 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 09:25 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
======================================================================
----------------------------------------------------------------------
(0118834) notthematrix (reporter) - 2010-03-03 09:25
https://issues.asterisk.org/view.php?id=5413#c118834
----------------------------------------------------------------------
@twilson ,here some report.
No still white noise. with SVN-group-srtp_reboot-r250289-/trunk
The grandstream ht-503 is in SRTP forced mode so it sould always request a
srtp encrypted call when placing a call.
When I played with the old 1.6.2 rc3 patch I had the same problem.
channel asterisk to grandstream (speaker) was white noise and channel
grandstream (mike) to asterisk worked oke.
we have it working properly with version SVN-group-srtp-r176603-/trunk
so tere might be something changed what is not liked.
Best regards Frank
here a working sip debug. of SVN-group-srtp-r176603-/trunk
<--- SIP read from TLS:92.254.1.100:2050 --->
INVITE sip:*66 at 87.222.43.11:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54261;branch=z9hG4bK1448223259;rport;alias
From: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
To: <sip:*66 at 87.222.43.11:443;user=phone>
Call-ID: 2074233693-54261-2 at 192.168.1.108
CSeq: 10 INVITE
Contact: <sip:788103 at 192.168.1.108:54261;transport=tls;user=phone>
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:788103 at 87.222.43.11:443;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=788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 44802 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:jov/A+BXlrWC54k48n7UUQ5RLymQhWlWOOzWff+R
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:r40cr5D8BiayiA07wP+6lFDI5X/xdQRazDxHorpG
<------------->
--- (16 headers 22 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 92.254.1.100 : 2050 (NAT)
Using INVITE request as basis request - 2074233693-54261-2 at 192.168.1.108
Found peer '788103' for '788103' from 92.254.1.100:2050
mastermetals2*CLI>
<--- Reliably Transmitting (NAT) to 92.254.1.100:2050 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/TLS
192.168.1.108:54261;branch=z9hG4bK1448223259;alias;received=92.254.1.100;rport=2050
f: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
t: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as3801be1b
i: 2074233693-54261-2 at 192.168.1.108
CSeq: 10 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="027011d3"
l: 0
<------------>
Scheduling destruction of SIP dialog '2074233693-54261-2 at 192.168.1.108' in
128000 ms (Method: INVITE)
mastermetals2*CLI>
<--- SIP read from TLS:92.254.1.100:2050 --->
ACK sip:*66 at 87.222.43.11:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54261;branch=z9hG4bK1448223259;rport;alias
From: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
To: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as3801be1b
Call-ID: 2074233693-54261-2 at 192.168.1.108
CSeq: 10 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
mastermetals2*CLI>
<--- SIP read from TLS:92.254.1.100:2050 --->
INVITE sip:*66 at 87.222.43.11:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54261;branch=z9hG4bK638909050;rport;alias
From: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
To: <sip:*66 at 87.222.43.11:443;user=phone>
Call-ID: 2074233693-54261-2 at 192.168.1.108
CSeq: 11 INVITE
Contact: <sip:788103 at 192.168.1.108:54261;transport=tls;user=phone>
Authorization: Digest username="788103", realm="asterisk",
nonce="027011d3", uri="sip:*66 at 87.222.43.11:443;user=phone",
response="fbff718527e37c840a040636d6010563", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:788103 at 87.222.43.11:443;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=788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 44802 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:jov/A+BXlrWC54k48n7UUQ5RLymQhWlWOOzWff+R
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:r40cr5D8BiayiA07wP+6lFDI5X/xdQRazDxHorpG
<------------->
--- (17 headers 22 lines) ---
Sending to 92.254.1.100 : 2050 (NAT)
Using INVITE request as basis request - 2074233693-54261-2 at 192.168.1.108
Found peer '788103' for '788103' from 92.254.1.100:2050
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
Peer audio RTP is at port 192.168.1.108:44802
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 192.168.1.108:44802
Looking for *66 in klant-31-123-123456 (domain 87.222.43.11)
list_route: hop:
<sip:788103 at 192.168.1.108:54261;transport=tls;user=phone>
<--- Transmitting (NAT) to 92.254.1.100:2050 --->
SIP/2.0 100 Trying
v: SIP/2.0/TLS
192.168.1.108:54261;branch=z9hG4bK638909050;alias;received=92.254.1.100;rport=2050
f: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
t: <sip:*66 at 87.222.43.11:443;user=phone>
i: 2074233693-54261-2 at 192.168.1.108
CSeq: 11 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces, timer
m: <sip:*66 at 87.222.43.11:443;transport=TLS>
l: 0
<------------>
-- Executing [*66 at klant-31-123-123456:1] Goto("SIP/788103-040bd5c8",
"from-internal,*66,1") in new stack
-- Goto (from-internal,*66,1)
-- Executing [*66 at from-internal:1] Answer("SIP/788103-040bd5c8", "")
in new stack
Audio is at 87.222.43.11 port 12552
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
<--- Reliably Transmitting (NAT) to 92.254.1.100:2050 --->
SIP/2.0 200 OK
v: SIP/2.0/TLS
192.168.1.108:54261;branch=z9hG4bK638909050;alias;received=92.254.1.100;rport=2050
f: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
t: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as20e26413
i: 2074233693-54261-2 at 192.168.1.108
CSeq: 11 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces, timer
m: <sip:*66 at 87.222.43.11:443;transport=TLS>
c: application/sdp
l: 455
v=0
o=root 1644478234 1644478234 IN IP4 87.222.43.11
s=Asterisk PBX
c=IN IP4 87.222.43.11
t=0 0
m=audio 12552 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:atQZFygtW/TTsGpFZFv/eX2MCnQN60FyibA+XAct
<------------>
mastermetals2*CLI>
<--- SIP read from TLS:92.254.1.100:2050 --->
ACK sip:*66 at 87.222.43.11:443;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54261;branch=z9hG4bK75050974;rport
From: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
To: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as20e26413
Call-ID: 2074233693-54261-2 at 192.168.1.108
CSeq: 11 ACK
Contact: <sip:788103 at 192.168.1.108:54261;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/788103-040bd5c8", "1") in
new stack
-- Executing [*66 at from-internal:3] Macro("SIP/788103-040bd5c8",
"user-callerid,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/788103-040bd5c8",
"AMPUSER=788103") in new stack
-- Executing [s at macro-user-callerid:2] GotoIf("SIP/788103-040bd5c8",
"0?report") in new stack
-- Executing [s at macro-user-callerid:3] ExecIf("SIP/788103-040bd5c8",
"1?Set(REALCALLERIDNUM=788103)") in new stack
-- Executing [s at macro-user-callerid:4] Set("SIP/788103-040bd5c8",
"AMPUSER=788103") in new stack
-- Executing [s at macro-user-callerid:5] Set("SIP/788103-040bd5c8",
"AMPUSERCIDNAME=test") in new stack
-- Executing [s at macro-user-callerid:6] GotoIf("SIP/788103-040bd5c8",
"0?report") in new stack
-- Executing [s at macro-user-callerid:7] Set("SIP/788103-040bd5c8",
"AMPUSERCID=788103") in new stack
-- Executing [s at macro-user-callerid:8] Set("SIP/788103-040bd5c8",
"CALLERID(all)="test" <788103>") in new stack
-- Executing [s at macro-user-callerid:9] GotoIf("SIP/788103-040bd5c8",
"0?continue") in new stack
-- Executing [s at macro-user-callerid:10] Set("SIP/788103-040bd5c8",
"__TTL=64") in new stack
-- Executing [s at macro-user-callerid:11] GotoIf("SIP/788103-040bd5c8",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s at macro-user-callerid:18] NoOp("SIP/788103-040bd5c8",
"Using CallerID "test" <788103>") in new stack
-- Executing [*66 at from-internal:4] DBdel("SIP/788103-040bd5c8",
"CP/788103") in new stack
-- DBdel: family=CP, key=788103
-- DBdel: Error deleting key from database.
-- Executing [*66 at from-internal:5] Playback("SIP/788103-040bd5c8",
"privacy-your-callerid-is&activated") in new stack
-- <SIP/788103-040bd5c8> Playing 'privacy-your-callerid-is.alaw'
(language 'en')
-- <SIP/788103-040bd5c8> Playing 'activated.alaw' (language 'en')
-- Executing [*66 at from-internal:6] Macro("SIP/788103-040bd5c8",
"hangupcall,") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/788103-040bd5c8",
"w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/788103-040bd5c8", "")
in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/788103-040bd5c8",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/788103-040bd5c8",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/788103-040bd5c8",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11] Hangup("SIP/788103-040bd5c8", "")
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/788103-040bd5c8' in macro 'hangupcall'
== Spawn extension (from-internal, *66, 6) exited non-zero on
'SIP/788103-040bd5c8'
-- Executing [h at from-internal:1] Macro("SIP/788103-040bd5c8",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/788103-040bd5c8",
"w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/788103-040bd5c8", "")
in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/788103-040bd5c8",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/788103-040bd5c8",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/788103-040bd5c8",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11] Hangup("SIP/788103-040bd5c8", "")
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/788103-040bd5c8' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/788103-040bd5c8'
Scheduling destruction of SIP dialog '2074233693-54261-2 at 192.168.1.108' in
128000 ms (Method: ACK)
set_destination: Parsing
<sip:788103 at 192.168.1.108:54261;transport=tls;user=phone> for address/port
to send to
set_destination: set destination to 192.168.1.108, port 54261
Reliably Transmitting (NAT) to 92.254.1.100:2050:
BYE sip:788103 at 192.168.1.108:54261;transport=tls;user=phone SIP/2.0
v: SIP/2.0/TLS 87.222.43.11:443;branch=z9hG4bK0e42916a;rport
Max-Forwards: 70
f: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as20e26413
t: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
i: 2074233693-54261-2 at 192.168.1.108
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
l: 0
---
mastermetals2*CLI>
<--- SIP read from TLS:92.254.1.100:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.222.43.11:443;branch=z9hG4bK0e42916a;rport=443
From: <sip:*66 at 87.222.43.11:443;user=phone>;tag=as20e26413
To: <sip:788103 at 87.222.43.11:443;user=phone>;tag=613547299
Call-ID: 2074233693-54261-2 at 192.168.1.108
CSeq: 102 BYE
Contact: <sip:788103 at 192.168.1.108:54261;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
Only ip nubers and (788103) was diffrent number , to anonymize the log.
here the whitenoise log , of the current version
SVN-group-srtp_reboot-r250289
OPTIONS sip:788103 at 192.168.1.108:54589;transport=tls;user=phone SIP/2.0
v: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK3e226472;rport
Max-Forwards: 70
f: "Unknown" <sip:Unknown at 87.222.43.12:443>;tag=as654ff079
t: <sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>
m: <sip:Unknown at 87.222.43.12:443;transport=TLS>
i: 047baddc5ec3c7be0044e6f64f363021 at 87.222.43.12
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 03 Mar 2010 01:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0
---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK3e226472;rport=443
From: "Unknown" <sip:Unknown at 87.222.43.12:443>;tag=as654ff079
To:
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>;tag=273863356
Call-ID: 047baddc5ec3c7be0044e6f64f363021 at 87.222.43.12
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:92.254.11.254:2050 --->
REGISTER sip:87.222.43.12:443 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK993758837;rport;alias
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1347247709
To: <sip:788103 at 87.222.43.12:443;user=phone>
Call-ID: 864877769-54589-1 at 192.168.1.108
CSeq: 2046 REGISTER
Contact:
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B82131B12>"
Authorization: Digest username="788103", realm="asterisk",
nonce="71a06e64", uri="sip:87.222.43.12:443",
response="149ab028eef9b4253cb764058fefde74", 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 92.254.11.254:2050 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/TLS
192.168.1.108:54589;branch=z9hG4bK993758837;alias;received=92.254.11.254;rport=2050
f: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1347247709
t: <sip:788103 at 87.222.43.12:443;user=phone>;tag=as2bd3f71a
i: 864877769-54589-1 at 192.168.1.108
CSeq: 2046 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="4566cd50"
l: 0
<------------>
Scheduling destruction of SIP dialog '864877769-54589-1 at 192.168.1.108' in
32000 ms (Method: REGISTER)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
REGISTER sip:87.222.43.12:443 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK2047545965;rport;alias
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1347247709
To: <sip:788103 at 87.222.43.12:443;user=phone>
Call-ID: 864877769-54589-1 at 192.168.1.108
CSeq: 2047 REGISTER
Contact:
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B82131B12>"
Authorization: Digest username="788103", realm="asterisk",
nonce="4566cd50", uri="sip:87.222.43.12:443",
response="4034bec2c25737a4e56a80fe5a8f2dae", 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 92.254.11.254 : 2050 (NAT)
Reliably Transmitting (NAT) to 92.254.11.254:2050:
OPTIONS sip:788103 at 192.168.1.108:54589;transport=tls;user=phone SIP/2.0
v: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK1cd63b4a;rport
Max-Forwards: 70
f: "Unknown" <sip:Unknown at 87.222.43.12:443>;tag=as6f2c96ef
t: <sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>
m: <sip:Unknown at 87.222.43.12:443;transport=TLS>
i: 13eeecb256b362f8433cf0e461ab2b19 at 87.222.43.12
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 03 Mar 2010 01:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0
---
mastermetals1*CLI>
<--- Transmitting (NAT) to 92.254.11.254:2050 --->
SIP/2.0 200 OK
v: SIP/2.0/TLS
192.168.1.108:54589;branch=z9hG4bK2047545965;alias;received=92.254.11.254;rport=2050
f: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1347247709
t: <sip:788103 at 87.222.43.12:443;user=phone>;tag=as2bd3f71a
i: 864877769-54589-1 at 192.168.1.108
CSeq: 2047 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
Expires: 120
m: <sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>;expires=120
Date: Wed, 03 Mar 2010 01:47:57 GMT
l: 0
<------------>
Scheduling destruction of SIP dialog '864877769-54589-1 at 192.168.1.108' in
32000 ms (Method: REGISTER)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK1cd63b4a;rport=443
From: "Unknown" <sip:Unknown at 87.222.43.12:443>;tag=as6f2c96ef
To:
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>;tag=701204345
Call-ID: 13eeecb256b362f8433cf0e461ab2b19 at 87.222.43.12
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) ---
Really destroying SIP dialog
'047baddc5ec3c7be0044e6f64f363021 at 87.222.43.12' Method: OPTIONS
Really destroying SIP dialog
'13eeecb256b362f8433cf0e461ab2b19 at 87.222.43.12' Method: OPTIONS
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
INVITE sip:*68 at 87.222.43.12:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK1734955060;rport;alias
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
To: <sip:*68 at 87.222.43.12:443;user=phone>
Call-ID: 229730171-54589-7 at 192.168.1.108
CSeq: 60 INVITE
Contact: <sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:788103 at 87.222.43.12:443;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=788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 53432 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:6lXR6Qk1NqvSrpH53b8cvei9h84XgPbQRRGaU82U
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:arfpO6HzcNeeQoUvO2LuVx/XFKalLCac/Gytlr96
<------------->
--- (16 headers 22 lines) ---
Using INVITE request as basis request - 229730171-54589-7 at 192.168.1.108
Found peer '788103' for '788103' from 92.254.11.254:2050
mastermetals1*CLI>
<--- Reliably Transmitting (NAT) to 92.254.11.254:2050 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/TLS
192.168.1.108:54589;branch=z9hG4bK1734955060;alias;received=92.254.11.254;rport=2050
f: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
t: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as6adf1547
i: 229730171-54589-7 at 192.168.1.108
CSeq: 60 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="0ded9599"
l: 0
<------------>
Scheduling destruction of SIP dialog '229730171-54589-7 at 192.168.1.108' in
6400 ms (Method: INVITE)
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
ACK sip:*68 at 87.222.43.12:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK1734955060;rport;alias
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
To: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as6adf1547
Call-ID: 229730171-54589-7 at 192.168.1.108
CSeq: 60 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
INVITE sip:*68 at 87.222.43.12:443;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK668750409;rport;alias
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
To: <sip:*68 at 87.222.43.12:443;user=phone>
Call-ID: 229730171-54589-7 at 192.168.1.108
CSeq: 61 INVITE
Contact: <sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>
Authorization: Digest username="788103", realm="asterisk",
nonce="0ded9599", uri="sip:*68 at 87.222.43.12:443;user=phone",
response="e8243ed929c8905de14c524b43626273", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.1.57
Privacy: none
P-Asserted-Identity: <sip:788103 at 87.222.43.12:443;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=788103 8000 8000 IN IP4 192.168.1.108
s=SIP Call
c=IN IP4 192.168.1.108
t=0 0
m=audio 53432 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:6lXR6Qk1NqvSrpH53b8cvei9h84XgPbQRRGaU82U
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:arfpO6HzcNeeQoUvO2LuVx/XFKalLCac/Gytlr96
<------------->
--- (17 headers 22 lines) ---
Sending to 92.254.11.254 : 2050 (NAT)
Using INVITE request as basis request - 229730171-54589-7 at 192.168.1.108
Found peer '788103' for '788103' from 92.254.11.254:2050
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 192.168.1.108:53432
Looking for *68 in klant-31-123-123456 (domain 87.222.43.12)
list_route: hop:
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone>
mastermetals1*CLI>
<--- Transmitting (NAT) to 92.254.11.254:2050 --->
SIP/2.0 100 Trying
v: SIP/2.0/TLS
192.168.1.108:54589;branch=z9hG4bK668750409;alias;received=92.254.11.254;rport=2050
f: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
t: <sip:*68 at 87.222.43.12:443;user=phone>
i: 229730171-54589-7 at 192.168.1.108
CSeq: 61 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
m: <sip:*68 at 87.222.43.12:443;transport=TLS>
l: 0
<------------>
-- Executing [*68 at klant-31-123-123456:1] Goto("SIP/788103-00000005",
"from-internal,*68,1") in new stack
-- Goto (from-internal,*68,1)
-- Executing [*68 at from-internal:1] NoOp("SIP/788103-00000005", "Bridge
signaling: ") in new stack
-- Executing [*68 at from-internal:2] NoOp("SIP/788103-00000005", "Bridge
media: ") in new stack
-- Executing [*68 at from-internal:3] Wait("SIP/788103-00000005", "1") in
new stack
-- Executing [*68 at from-internal:4] Answer("SIP/788103-00000005", "")
in new stack
Audio is at 87.222.43.12 port 19784
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 92.254.11.254:2050 --->
SIP/2.0 200 OK
v: SIP/2.0/TLS
192.168.1.108:54589;branch=z9hG4bK668750409;alias;received=92.254.11.254;rport=2050
f: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
t: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as08af6e14
i: 229730171-54589-7 at 192.168.1.108
CSeq: 61 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
m: <sip:*68 at 87.222.43.12:443;transport=TLS>
c: application/sdp
l: 457
v=0
o=root 2136709363 2136709363 IN IP4 87.222.43.12
s=Asterisk PBX
c=IN IP4 87.222.43.12
t=0 0
m=audio 19784 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:YXreDqKfG+DLsPDdtKES5z6pSCkbb6WKVLW+17ng
<------------>
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
ACK sip:*68 at 87.222.43.12:443;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.108:54589;branch=z9hG4bK240260884;rport
From: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
To: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as08af6e14
Call-ID: 229730171-54589-7 at 192.168.1.108
CSeq: 61 ACK
Contact: <sip:788103 at 192.168.1.108:54589;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 [*68 at from-internal:5] Macro("SIP/788103-00000005",
"user-callerid,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/788103-00000005",
"AMPUSER=788103") in new stack
-- Executing [s at macro-user-callerid:2] GotoIf("SIP/788103-00000005",
"0?report") in new stack
-- Executing [s at macro-user-callerid:3] ExecIf("SIP/788103-00000005",
"1?Set(REALCALLERIDNUM=788103)") in new stack
-- Executing [s at macro-user-callerid:4] Set("SIP/788103-00000005",
"AMPUSER=") in new stack
-- Executing [s at macro-user-callerid:5] Set("SIP/788103-00000005",
"AMPUSERCIDNAME=") in new stack
-- Executing [s at macro-user-callerid:6] GotoIf("SIP/788103-00000005",
"1?report") in new stack
-- Goto (macro-user-callerid,s,9)
-- Executing [s at macro-user-callerid:9] GotoIf("SIP/788103-00000005",
"0?continue") in new stack
-- Executing [s at macro-user-callerid:10] Set("SIP/788103-00000005",
"__TTL=64") in new stack
-- Executing [s at macro-user-callerid:11] GotoIf("SIP/788103-00000005",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s at macro-user-callerid:18] NoOp("SIP/788103-00000005",
"Using CallerID "device" <788103>") in new stack
-- Executing [*68 at from-internal:6] Set("SIP/788103-00000005",
"DB(CP/)=ENABLED") in new stack
-- Executing [*68 at from-internal:7] Playback("SIP/788103-00000005",
"privacy-your-callerid-is&de-activated") in new stack
-- <SIP/788103-00000005> Playing 'privacy-your-callerid-is.alaw'
(language 'en')
-- <SIP/788103-00000005> Playing 'de-activated.alaw' (language 'en')
-- Executing [*68 at from-internal:8] Macro("SIP/788103-00000005",
"hangupcall,") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/788103-00000005",
"w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/788103-00000005", "")
in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/788103-00000005",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/788103-00000005",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/788103-00000005",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11] Hangup("SIP/788103-00000005", "")
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/788103-00000005' in macro 'hangupcall'
== Spawn extension (from-internal, *68, 8) exited non-zero on
'SIP/788103-00000005'
-- Executing [h at from-internal:1] Macro("SIP/788103-00000005",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/788103-00000005",
"w") in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/788103-00000005", "")
in new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/788103-00000005",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/788103-00000005",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/788103-00000005",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11] Hangup("SIP/788103-00000005", "")
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/788103-00000005' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/788103-00000005'
Scheduling destruction of SIP dialog '229730171-54589-7 at 192.168.1.108' in
6400 ms (Method: INVITE)
set_destination: Parsing
<sip:788103 at 192.168.1.108:54589;transport=tls;user=phone> for address/port
to send to
set_destination: set destination to 192.168.1.108, port 54589
Reliably Transmitting (NAT) to 92.254.11.254:2050:
BYE sip:788103 at 192.168.1.108:54589;transport=tls;user=phone SIP/2.0
v: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK0659ee6d;rport
Max-Forwards: 70
f: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as08af6e14
t: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
i: 229730171-54589-7 at 192.168.1.108
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="788103", realm="asterisk",
algorithm=MD5, uri="87.222.43.12", nonce="",
response="36cb4d85b80aec9a89c3e3eb1736be73"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
l: 0
---
mastermetals1*CLI>
<--- SIP read from TLS:92.254.11.254:2050 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 87.222.43.12:443;branch=z9hG4bK0659ee6d;rport=443
From: <sip:*68 at 87.222.43.12:443;user=phone>;tag=as08af6e14
To: <sip:788103 at 87.222.43.12:443;user=phone>;tag=1797763342
Call-ID: 229730171-54589-7 at 192.168.1.108
CSeq: 102 BYE
Contact: <sip:788103 at 192.168.1.108:54589;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
Issue History
Date Modified Username Field Change
======================================================================
2010-03-03 09:25 notthematrix Note Added: 0118834
======================================================================
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