[jitsi-users] client firewall


#1

Hi,

I'm trying out a new softphone and I'm really impressed by the "features" of JITSI (www.jitsi.org) even though it's still in beta.

Precisely because it's in beta I'm having trouble making calls through Asterisk. I'd like to know if someone here has managed to make it work.

I'm using Jitsi 1.0 beta1 build 3442 (32-bit x86 Windows binary).
I can register to Asterisk just fine as <exten>@<asterisk_IP_addr> ("sip show peers" tells me that my monitored extension is online).
I can receive calls routed from Asterisk (Jitsi rings and I can talk to caller).
However, I cannot dial out. I try to contact <other_exten>@<asterisk_IP_addr> (or simply call back the last caller in my Jitsi History) but Jitsi doesn't go beyond "Initiating call".

It's a Windows 7 firewall issue (where Jitsi is running) because if I fully disable Win Firewall then I CAN dial out correctly.

However, I'd like to leave Windows firewall ON and it already has the following exceptions:
C:\program files\jitsi\jre\bin\javaw.exe
C:\program files\jitsi\run.exe

If I set sip debug in Asterisk then this is what I get when trying to call from my Jitsi softphone at 10.215.144.48 registered as 4053 at asterisk server 10.215.145.112 to extension 3210:

srv-voip2*CLI>
<--- SIP read from 10.215.144.48:5060 --->
INVITE sip:3210@10.215.147.112 SIP/2.0
Call-ID: 8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0
CSeq: 1 INVITE

From: "4053" <sip:4053@10.215.147.112>;tag=556f2285

Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66
Max-Forwards: 70
User-Agent: Jitsi1.0-beta1-nightly.build.3442Windows 7
Content-Type: application/sdp
Content-Length: 841

v=0
o=4053 0 0 IN IP4 10.215.144.48
s=-
c=IN IP4 10.215.144.48
t=0 0
m=audio 5008 RTP/AVP 9 96 97 0 8 98 3 99 5 6 4 15 101
a=rtpmap:9 G722/8000
a=rtpmap:96 speex/32000
a=rtpmap:97 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 speex/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:6 DVI4/16000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:15 G728/8000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=zrtp-hash:1.10 996db3393c53f46cf624e88b1d25ff9cf3b0fd0f11555dbf931b42b141b7e89a
m=video 5010 RTP/AVP 100
a=recvonly
a=rtpmap:100 H264/90000
a=fmtp:100 packetization-mode=1
a=imageattr:100 send * recv [x=[0-1280],y=[0-1024]]
a=zrtp-hash:1.10 fbcd82761d8479d025a55ae39a87c0fe942e0801658670228ea1c5915e80dd87

<------------->
--- (11 headers 28 lines) ---
Sending to 10.215.144.48 : 5060 (no NAT)
Using INVITE request as basis request - 8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0
srv-voip2*CLI>
<--- Reliably Transmitting (NAT) to 10.215.144.48:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=556f2285

Call-ID: 8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="485351d4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0' in 32000 ms (Method: INVITE)
Found user '4053'
srv-voip2*CLI>
<--- SIP read from 10.215.144.48:5060 --->
ACK sip:3210@10.215.147.112 SIP/2.0
Call-ID: 8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0
Max-Forwards: 70

From: "4053" <sip:4053@10.215.147.112>;tag=556f2285

Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66
CSeq: 1 ACK
Content-Length: 0

<------------->
srv-voip2*CLI>
<--- SIP read from 10.215.144.48:5060 --->
REGISTER sip:10.215.147.112 SIP/2.0
Call-ID: 03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0
CSeq: 39 REGISTER

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d

Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c
Max-Forwards: 70
Authorization: Digest username="4053",realm="asterisk",nonce="02077fa4",uri="sip:10.215.147.112",response="ed0a4b2fe02d6b24831ece74ea39817a",algorithm=MD5
User-Agent: Jitsi1.0-beta1-nightly.build.3442Windows 7
Expires: 600
Contact: "4053" <sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>;expires=600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.215.144.48 : 5060 (NAT)
srv-voip2*CLI>
<--- Transmitting (NAT) to 10.215.144.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d

Call-ID: 03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0
CSeq: 39 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
srv-voip2*CLI>
<--- Transmitting (NAT) to 10.215.144.48:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d

Call-ID: 03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0
CSeq: 39 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06d4d3d7", stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0' in 32000 ms (Method: REGISTER)
srv-voip2*CLI>
<--- SIP read from 10.215.144.48:5060 --->
REGISTER sip:10.215.147.112 SIP/2.0
Call-ID: 03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0
CSeq: 40 REGISTER

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d

Max-Forwards: 70
User-Agent: Jitsi1.0-beta1-nightly.build.3442Windows 7
Expires: 600
Contact: "4053" <sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>;expires=600
Via: SIP/2.0/UDP 10.215.144.48:5060;branch=z9hG4bK-393531-edbee168c8b2eadac6145d74e244b36e
Authorization: Digest username="4053",realm="asterisk",nonce="06d4d3d7",uri="sip:10.215.147.112",response="0a58a61b909b8f5652e823f39a02599c",algorithm=MD5
Content-Length: 0

What do I need to change in my Windows 7 firewall so that Jitsi can dial out?

Also, whether I disable Windows firewall or not, I get a windows crash message when launching Jitsi. It seems to happen when run.exe is executed. The Java app appears and seems to work but I get an APPCRASH on run.exe (module with errors: ntdll.dll).

I'm supposing this is a different issue.

Thanks,

Vieri

···

To: <sip:3210@10.215.147.112>
Contact: "4053" <sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>
To: <sip:3210@10.215.147.112>;tag=as78d98592
To: <sip:3210@10.215.147.112>;tag=as78d98592
To: "4053" <sip:4053@10.215.147.112>
To: "4053" <sip:4053@10.215.147.112>
To: "4053" <sip:4053@10.215.147.112>;tag=as091e07d4
To: "4053" <sip:4053@10.215.147.112>


#2

Also, whether I disable Windows firewall or not, I get a windows crash message when launching Jitsi. It seems to happen when run.exe is executed. The Java app appears and seems to work but I get an APPCRASH on run.exe (module with errors: ntdll.dll).

I'm supposing this is a different issue.

Yes, this is a different issue.

Could you please provide more details with respect to the run.exe
crash? For example, any hs_err_pid*.log files in "C:\Program
Files\Jitsi" or %TEMP%?


#3

Hey Vieri,

На 27.04.11 16:32, Vieri написа:

Hi,

I'm trying out a new softphone and I'm really impressed by the
"features" of JITSI (www.jitsi.org) even though it's still in beta.

Precisely because it's in beta I'm having trouble making calls
through Asterisk. I'd like to know if someone here has managed to
make it work.

I know of many people that are actually. I am personally making tens of
audio and video calls through asterisk with Jitsi every day and don't
have any issues.

I can't be sure about the issue from these log files but could you
please try deleting and creating your account again?

If the problem persists, could you please send us the log zip [0] ?

Cheers,
Emil

[0] http://jitsi.org/faq/logs

I'm using Jitsi 1.0 beta1 build 3442 (32-bit x86 Windows binary). I
can register to Asterisk just fine as <exten>@<asterisk_IP_addr>
("sip show peers" tells me that my monitored extension is online). I
can receive calls routed from Asterisk (Jitsi rings and I can talk to
caller). However, I cannot dial out. I try to contact
<other_exten>@<asterisk_IP_addr> (or simply call back the last caller
in my Jitsi History) but Jitsi doesn't go beyond "Initiating call".

It's a Windows 7 firewall issue (where Jitsi is running) because if I
fully disable Win Firewall then I CAN dial out correctly.

However, I'd like to leave Windows firewall ON and it already has the
following exceptions: C:\program files\jitsi\jre\bin\javaw.exe
C:\program files\jitsi\run.exe

If I set sip debug in Asterisk then this is what I get when trying to
call from my Jitsi softphone at 10.215.144.48 registered as 4053 at
asterisk server 10.215.145.112 to extension 3210:

srv-voip2*CLI> <--- SIP read from 10.215.144.48:5060 ---> INVITE
sip:3210@10.215.147.112 SIP/2.0 Call-ID:
8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0 CSeq: 1 INVITE From:
"4053" <sip:4053@10.215.147.112>;tag=556f2285 To:
<sip:3210@10.215.147.112> Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66

Max-Forwards: 70

Contact: "4053"
<sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>

User-Agent: Jitsi1.0-beta1-nightly.build.3442Windows 7

Content-Type: application/sdp Content-Length: 841

v=0 o=4053 0 0 IN IP4 10.215.144.48 s=- c=IN IP4 10.215.144.48 t=0 0
m=audio 5008 RTP/AVP 9 96 97 0 8 98 3 99 5 6 4 15 101 a=rtpmap:9
G722/8000 a=rtpmap:96 speex/32000 a=rtpmap:97 speex/16000 a=rtpmap:0
PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:3
GSM/8000 a=rtpmap:99 speex/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:6
DVI4/16000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:15 G728/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1
urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10
996db3393c53f46cf624e88b1d25ff9cf3b0fd0f11555dbf931b42b141b7e89a
m=video 5010 RTP/AVP 100 a=recvonly a=rtpmap:100 H264/90000
a=fmtp:100 packetization-mode=1 a=imageattr:100 send * recv
[x=[0-1280],y=[0-1024]] a=zrtp-hash:1.10
fbcd82761d8479d025a55ae39a87c0fe942e0801658670228ea1c5915e80dd87

<-------------> --- (11 headers 28 lines) --- Sending to
10.215.144.48 : 5060 (no NAT) Using INVITE request as basis request -
8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0 srv-voip2*CLI> <---
Reliably Transmitting (NAT) to 10.215.144.48:5060 ---> SIP/2.0 407
Proxy Authentication Required Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=556f2285
To: <sip:3210@10.215.147.112>;tag=as78d98592 Call-ID:
8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0 CSeq: 1 INVITE
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="485351d4" Content-Length: 0

<------------> Scheduling destruction of SIP dialog
'8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0' in 32000 ms
(Method: INVITE) Found user '4053' srv-voip2*CLI> <--- SIP read from
10.215.144.48:5060 ---> ACK sip:3210@10.215.147.112 SIP/2.0 Call-ID:
8cfe8d9722d8a59c5816da43e710379c@0:0:0:0:0:0:0:0 Max-Forwards: 70
From: "4053" <sip:4053@10.215.147.112>;tag=556f2285 To:
<sip:3210@10.215.147.112>;tag=as78d98592 Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-eebadee2eae8be9dfa4f04cedef33a66

CSeq: 1 ACK

Content-Length: 0

<-------------> srv-voip2*CLI> <--- SIP read from 10.215.144.48:5060
---> REGISTER sip:10.215.147.112 SIP/2.0 Call-ID:
03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0 CSeq: 39 REGISTER
From: "4053" <sip:4053@10.215.147.112>;tag=babc004d To: "4053"
<sip:4053@10.215.147.112> Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c

Max-Forwards: 70

Authorization: Digest
username="4053",realm="asterisk",nonce="02077fa4",uri="sip:10.215.147.112",response="ed0a4b2fe02d6b24831ece74ea39817a",algorithm=MD5

User-Agent: Jitsi1.0-beta1-nightly.build.3442Windows 7

Expires: 600 Contact: "4053"
<sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>;expires=600

Content-Length: 0

<-------------> --- (12 headers 0 lines) --- Using latest REGISTER
request as basis request Sending to 10.215.144.48 : 5060 (NAT)
srv-voip2*CLI> <--- Transmitting (NAT) to 10.215.144.48:5060 --->
SIP/2.0 100 Trying Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d
To: "4053" <sip:4053@10.215.147.112> Call-ID:
03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0 CSeq: 39 REGISTER
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0

<------------> srv-voip2*CLI> <--- Transmitting (NAT) to
10.215.144.48:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-38e4a202d14efe8c492bb0a49190fa6c;received=10.215.144.48

From: "4053" <sip:4053@10.215.147.112>;tag=babc004d
To: "4053" <sip:4053@10.215.147.112>;tag=as091e07d4 Call-ID:
03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0 CSeq: 39 REGISTER
User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate:
Digest algorithm=MD5, realm="asterisk", nonce="06d4d3d7", stale=true
Content-Length: 0

<------------> Scheduling destruction of SIP dialog
'03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0' in 32000 ms
(Method: REGISTER) srv-voip2*CLI> <--- SIP read from
10.215.144.48:5060 ---> REGISTER sip:10.215.147.112 SIP/2.0 Call-ID:
03886d0e455189f93b5746cb78e7b7ed@0:0:0:0:0:0:0:0 CSeq: 40 REGISTER
From: "4053" <sip:4053@10.215.147.112>;tag=babc004d To: "4053"
<sip:4053@10.215.147.112> Max-Forwards: 70 User-Agent:
Jitsi1.0-beta1-nightly.build.3442Windows 7 Expires: 600 Contact:
"4053"
<sip:4053@10.215.144.48:5060;transport=udp;registering_acc=10_215_147_112>;expires=600

Via: SIP/2.0/UDP
10.215.144.48:5060;branch=z9hG4bK-393531-edbee168c8b2eadac6145d74e244b36e

Authorization: Digest
username="4053",realm="asterisk",nonce="06d4d3d7",uri="sip:10.215.147.112",response="0a58a61b909b8f5652e823f39a02599c",algorithm=MD5

Content-Length: 0

···

What do I need to change in my Windows 7 firewall so that Jitsi can
dial out?

Also, whether I disable Windows firewall or not, I get a windows
crash message when launching Jitsi. It seems to happen when run.exe
is executed. The Java app appears and seems to work but I get an
APPCRASH on run.exe (module with errors: ntdll.dll).

I'm supposing this is a different issue.

Thanks,

Vieri

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31


#4

I haven't found any such log files anywhere.

Just to make it clear: the run.exe crash message pops up BUT the Jitsi java application comes up after that and seems to work fine.

Vieri

···

--- On Wed, 4/27/11, Lyubomir Marinov <lubo@jitsi.org> wrote:

> Also, whether I disable Windows
firewall or not, I get a windows crash message when
launching Jitsi. It seems to happen when run.exe is
executed. The Java app appears and seems to work but I get
an APPCRASH on run.exe (module with errors: ntdll.dll).
>
> I'm supposing this is a different issue.

Yes, this is a different issue.

Could you please provide more details with respect to the
run.exe
crash? For example, any hs_err_pid*.log files in
"C:\Program
Files\Jitsi" or %TEMP%?