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<DIV><FONT face="MS UI Gothic">
<DIV><FONT face="MS UI Gothic">Hi </FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">I tryed to build CONFIANCE Video mixer and
confirm video mixed,<BR>refering to <A
href="http://confiance.sourceforge.net/">http://confiance.sourceforge.net/</A>,
especially Installer script.</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">I constructed the environment below.<BR> *
Asterisk(1.4.3) with CONFIANCE
patch(confiance_patch_asterisk_jun14th-07.diff)<BR> * CONFIANCE video
mixer(confiance_vm)<BR> * minisip(revision 2760) with CONFIANCE
patch(confiance_patch_minisip_jun14th-07.diff)<BR>But unfortunately it does not
work correct.</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">Could someone help me with this
problem?</FONT></DIV>
<DIV><FONT face="MS UI Gothic">Perhaps Lorenzo!</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">I show the infomation about our environment
bellow.</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">1. operation<BR>1.1 server<BR> (1) startup
video mixer<BR> confiance_vm 7000<BR> (2) startup
asterisk<BR> asterisk -gcvvvvvvvvvvv</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">1.2 minisip<BR> 1.2.1
startup<BR> # ./build.pl run minisip</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic"> 1.2.2 join an XCON
Conference<BR> I operate minisip GUI as shown below.<BR>
(1)enable and check XCON on "VIEW" menu bar.<BR> (2)input Confernce
number(8671000) to "Extension" input area on "XCON" tab window.<BR>
(3)push "join this XCON Conference" button on "XCON" tab window. </FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">Now I wonder how to use video conference on
minisip, because I can't find document about video confernece.<BR>Please tell me
correct operation about video confernece, if I mistake.</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">2. packet between minisip and asterisk<BR>After I
operate by the procedure above,<BR>I confirmed the session between minisip and
asterisk.<BR>But it seemed that only audio session was established, but video
session was not.<BR>After established session, I confirmed audio RTP packet ,but
not video RTP packet.</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">I show sip packet between minisip and asterisk
below.<BR>------------------SIP PACKET
FLOW----------------------------------<BR>(1) minisip(IPaddr:192.168.1.173)
-> asterisk(IPaddr:192.168.1.21)<BR> Request-Line: INVITE
sip:8671000@192.168.1.21 SIP/2.0<BR> Message
Header<BR> Route:
<sip:192.168.1.21:5060;transport=UDP;lr><BR>
From:
<sip:2001@192.168.1.21>;tag=1619364546<BR>
To:
<sip:8671000@192.168.1.21><BR>
Call-ID: <A
href="mailto:438183866@192.168.1.173">438183866@192.168.1.173</A><BR>
CSeq: 801 INVITE<BR> Contact:
<sip:2001@192.168.1.173:5060;transport=UDP>;expires=1000<BR>
User-Agent: Minisip<BR> Supported:
100rel<BR> Content-Type:
application/sdp<BR> Via: SIP/2.0/UDP
192.168.1.173:5060;rport;branch=z9hG4bK976758295<BR>
Content-Length: 357<BR> Message
body<BR> Session Description
Protocol<BR>
Session Description Protocol Version (v):
0<BR>
Owner/Creator, Session Id (o): - 3344 3344 IN IP4
192.168.1.173<BR>
Session Name (s): Minisip
Session<BR>
Time Description, active time (t): 0
0<BR> Media
Description, name and address (m): audio 30554 RTP/AVP 0
101<BR>
Connection Information (c): IN IP4
192.168.1.173<BR>
Media Attribute (a): rtpmap:0
PCMU/8000/1<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-15<BR> Media
Description, name and address (m): video 30992 RTP/AVP 105
101<BR>
Connection Information (c): IN IP4
192.168.1.173<BR>
Media Attribute (a): rtpmap:105
h263-1998/90000<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-15<BR> Media
Attribute (a): framesize:34 176-144<BR>(2) asterisk ->
minisip<BR> Status-Line: SIP/2.0 100 Trying</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">(3) asterisk ->
minisip<BR> Status-Line: SIP/2.0 200
OK<BR> Message
Header<BR> Via: SIP/2.0/UDP
192.168.1.173:5060;branch=z9hG4bK976758295;received=192.168.1.173;rport=5060<BR>
From:
<sip:2001@192.168.1.21>;tag=1619364546<BR>
To:
<sip:8671000@192.168.1.21>;tag=as446a6ab2<BR>
Call-ID: <A
href="mailto:438183866@192.168.1.173">438183866@192.168.1.173</A><BR>
CSeq: 801 INVITE<BR> User-Agent:
Asterisk PBX<BR> Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY<BR> Supported:
replaces<BR> Contact:
<sip:8671000@192.168.1.21><BR>
Content-Type: application/sdp<BR>
Content-Length: 240<BR> Message
body<BR> Session Description
Protocol<BR>
Session Description Protocol Version (v):
0<BR>
Owner/Creator, Session Id (o): root 11774 11774 IN IP4
192.168.1.21<BR>
Session Name (s):
session<BR>
Connection Information (c): IN IP4
192.168.1.21<BR>
Time Description, active time (t): 0
0<BR> Media
Description, name and address (m): audio 25836 RTP/AVP 0
101<BR> Media
Attribute (a): rtpmap:0
PCMU/8000<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-16<BR> Media
Attribute (a): silenceSupp:off - - -
-<BR> Media
Attribute (a):
ptime:20<BR>
Media Attribute (a): sendrecv</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">(4) minisip -> asterisk<BR>
Request-Line: ACK sip:8671000@192.168.1.21 SIP/2.0</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">(5) asterisk -> minisip<BR>
Request-Line: INVITE sip:2001@192.168.1.173:5060;transport=UDP
SIP/2.0<BR> Message
Header<BR> Via: SIP/2.0/UDP
192.168.1.21:5060;branch=z9hG4bK3b9cc93d;rport<BR>
From:
<sip:8671000@192.168.1.21>;tag=as446a6ab2<BR>
To:
<sip:2001@192.168.1.21>;tag=1619364546<BR>
Contact:
<sip:8671000@192.168.1.21><BR>
Call-ID: <A
href="mailto:438183866@192.168.1.173">438183866@192.168.1.173</A><BR>
CSeq: 102 INVITE<BR> User-Agent:
Asterisk PBX<BR> Max-Forwards:
70<BR> Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY<BR> Supported:
replaces<BR> Content-Type:
application/sdp<BR> Content-Length:
420<BR> Message
body<BR> Session Description
Protocol<BR>
Session Description Protocol Version (v):
0<BR>
Owner/Creator, Session Id (o): root 11774 11775 IN IP4
192.168.1.21<BR>
Session Name (s):
session<BR>
Connection Information (c): IN IP4
192.168.1.21<BR>
Time Description, active time (t): 0
0<BR> Media
Description, name and address (m): audio 25836 RTP/AVP 0
101<BR> Media
Attribute (a): rtpmap:0
PCMU/8000<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-16<BR> Media
Attribute (a): silenceSupp:off - - -
-<BR> Media
Attribute (a):
ptime:20<BR>
Media Attribute (a):
label:10<BR>
Media Attribute (a):
sendrecv<BR>
Media Description, name and address (m): application 2345 TCP/BFCP
*<BR> Media
Attribute (a):
setup:passive<BR>
Media Attribute (a):
connection:new<BR>
Media Attribute (a):
floorctrl:s-only<BR>
Media Attribute (a):
confid:8671000<BR>
Media Attribute (a):
userid:1<BR>
Media Attribute (a): floorid:11
m-stream:10<BR>
Media Attribute (a): floorid:22 m-stream:11</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">(6) minisip -> asterisk<BR>
Status-Line: SIP/2.0 200 OK<BR> Message
Header<BR> Max-Forwards:
70<BR> Via: SIP/2.0/UDP
192.168.1.21:5060;branch=z9hG4bK3b9cc93d;rport=5060<BR>
From:
<sip:8671000@192.168.1.21>;tag=as446a6ab2<BR>
To:
<sip:2001@192.168.1.21>;tag=1619364546<BR>
Call-ID: <A
href="mailto:438183866@192.168.1.173">438183866@192.168.1.173</A><BR>
CSeq: 102 INVITE<BR> Contact:
<sip:2001@192.168.1.173:5060;transport=UDP><BR>
Content-Type: application/sdp<BR>
Content-Length: 439<BR> Message
body<BR> Session Description
Protocol<BR>
Session Description Protocol Version (v):
0<BR>
Owner/Creator, Session Id (o): - 3344 3344 IN IP4
192.168.1.173<BR>
Session Name (s): Minisip
Session<BR>
Time Description, active time (t): 0
0<BR> Media
Description, name and address (m): audio 30554 RTP/AVP 0
101<BR>
Connection Information (c): IN IP4
192.168.1.173<BR>
Media Attribute (a): rtpmap:0
PCMU/8000/1<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-15<BR> Media
Description, name and address (m): video 30992 RTP/AVP 105
101<BR>
Connection Information (c): IN IP4
192.168.1.173<BR>
Media Attribute (a): rtpmap:105
h263-1998/90000<BR>
Media Attribute (a): rtpmap:101
telephone-event/8000<BR>
Media Attribute (a): fmtp:101
0-15<BR> Media
Attribute (a): framesize:34
176-144<BR>
Media Description, name and address (m): application 9 TCP/BFCP
*<BR> Media
Attribute (a):
setup:active<BR>
Media Attribute (a):
floorctrl:c-only<BR>
Media Attribute (a): connection:new</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">(7) asterisk -> minisip<BR>
Request-Line: ACK sip:2001@192.168.1.173:5060;transport=UDP
SIP/2.0<BR>----------------------------------------------------------------------------</FONT></DIV>
<DIV><FONT face="MS UI Gothic"></FONT> </DIV>
<DIV><FONT face="MS UI Gothic">3. Asterisk console log<BR>I show the console log
on Asterisk below, when I join XCON conference on minisip.</FONT></DIV>
<DIV><FONT face="MS UI Gothic">--------------------Asterisk console log(join
XCON conference)--------------<BR>*CLI> -- Executing
[8671000@default:1] MeetMe("SIP/2001-08f8aa20", "8671000|B|") in new
stack<BR> == Parsing '/etc/asterisk/xcon.conf': Found<BR>[Sep 13 10:29:33]
WARNING[11809]: config.c:756 process_text_line: No '=' (equal sign) in line 52
of /etc/asterisk/xcon.conf<BR> -- The new local conference
(ConferenceID: 8671000) has been added to the BFCP Server:<BR>
-- Floor: Audio, ID 11 (unlimited
users)<BR> -- Floor: Video, ID 22
(limited users)<BR> -- Adding
conference to the BFCP Server: DONE<BR> -- Created XCON
conference 1023 for conference '8671000'<BR> --
Requesting new VideoMixer session for conference 8671000<BR>
-- New Participant has UserID 1 (Conference 8671000)...<BR>
-- CallerID: , URI: sip:2001@192.168.1.173:5060<BR>[Sep 13 10:29:33]
WARNING[11809]: app_meetme.c:2841 conf_run: Couldn't add UserID 1 to Conference
8671000 Users' list...<BR> -- Sending required
BFCP+MSRP information to chan_sip...<BR>
-- BFCP information structure for SDP received from
MeetMe...<BR> -- Video Format: no video<BR>
-- [CVM] Conference 8671000 --> Session 1<BR> --
<SIP/2001-08f8aa20> Playing 'conf-onlyperson' (language
'en')<BR> -- ACK from XCON client received, requesting
reinvite...<BR> -- Transmitting pending reinvite
with BFCP information...<BR> --
Building SDP+BFCP/MSRP...<BR> -- Actually sending reinvite
with BFCP information...<BR> -- BFCP [8671000/1/1] <---
Hello<BR> -- BFCP [8671000/1/1] --->
HelloAck<BR> -- BFCP [8671000/2/1] <---
FloorQuery<BR> -- BFCP [8671000/2/1] --->
FloorStatus<BR> -- [XCON] XconScheduler:
QueryUsers<BR> -- [XCON] InfoUsers<BR>
-- Parsing BFCP information in SIP OK's SDP: TCP/BFCP (bfcp port = 9,
discard)...<BR> -- UserID 1 (Conference 8671000)
MUTED<BR> -- <SIP/2001-08f8aa20> Playing 'conf-muted'
(language 'en')<BR> -- Notifying AudioFloor change
to chan_sip...<BR> -- Not notifying
AudioFloor change to chan_sip, user just
joined...<BR>---------------------------------------------------------------------------</FONT></DIV>
<DIV><FONT face="MS UI Gothic">Regards,</FONT></DIV></FONT></DIV></BODY></HTML>