[sip-comm-dev] Problem with zrtp


#1

Hi,

I’m having a problem related to zrtp when I try to call a user, indeed when
SIPcomm is creating the invite request’s content. The problem is the
following:

java.net.MalformedURLException: invalid url:
reference:file:sc-bundles/zrtp4j.jar!/ (java.net.MalformedURLException:
Unknown protocol: reference)

     [java] at java.net.URL.<init>(URL.java:601)

     [java] at
org.apache.felix.framework.URLHandlersStreamHandlerProxy.parseURL(URLHandler
sStreamHandlerProxy.java:281)

     [java] at java.net.URL.<init>(URL.java:596)

     [java] at java.net.URL.<init>(URL.java:464)

     [java] at java.net.URL.<init>(URL.java:413)

     [java] at javax.crypto.SunJCE_c.b(DashoA13*..)

     [java] at javax.crypto.SunJCE_c.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_c.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_b.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_h.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_h.a(DashoA13*..)

     [java] at javax.crypto.Cipher.c(DashoA13*..)

     [java] at javax.crypto.Cipher.b(DashoA13*..)

     [java] at javax.crypto.Cipher.getInstance(DashoA13*..)

     [java] at gnu.java.zrtp.ZRtp.<init>(ZRtp.java:329)

     [java] at
net.java.sip.communicator.impl.media.transform.zrtp.ZRTPTransformEngine.init
ialize(ZRTPTransformEngine.java:447)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.initializeRtpManager(Ca
llSessionImpl.java:2123)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.allocateMediaPorts(Call
SessionImpl.java:2026)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.createSessionDescriptio
n(CallSessionImpl.java:1419)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.createSdpOffer(CallSess
ionImpl.java:646)

     [java] at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImp
l.createOutgoingCall(OperationSetBasicTelephonySipImpl.java:298)

     [java] at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImp
l.createCall(OperationSetBasicTelephonySipImpl.java:165)

     [java] at
net.java.sip.communicator.impl.gui.main.call.CallManager$CreateCallThread.ru
n(CallManager.java:377)

I don’t know what it is this problem, and what is more yesterday I didn’t
get this problem doing the same call. Can anybody help me, please?

Another thing else related to zrtp, How could I deactivate the zrtp
encryption? I’m working with a server which catch the sip invite message
and it don’t tolerate the following attribute related to zrtp.

Thanks a lot, and best regards

Pedro Antonio Moreno Sanchez
Telecommunication Engineer & Researcher
Bioengineering and Telemedicine Group
ETSI Telecomunicación - UPM B.302.
Ciudad Universitaria s/n
28040 MADRID - SPAIN

Tel. +34 91 549 57 00 ext. 3407
Fax + 34 91 336 68 28
e-mail: <mailto:pmoreno@gbt.tfo.upm.es> pmoreno@gbt.tfo.upm.es


#2

Pedro,

it seems that you use a fairly old version ;-). Can you update to the
latest nightly build and try again. ZRTP does not use the SunJCE
anymore, since about 2 or 3 weeks.

Regards,
Werner

Pedro Antonio Moreno Sanchez schrieb:

···

Hi,

I’m having a problem related to zrtp when I try to call a user, indeed when
SIPcomm is creating the invite request’s content. The problem is the
following:

java.net.MalformedURLException: invalid url:
reference:file:sc-bundles/zrtp4j.jar!/ (java.net.MalformedURLException:
Unknown protocol: reference)

     [java] at java.net.URL.<init>(URL.java:601)

     [java] at
org.apache.felix.framework.URLHandlersStreamHandlerProxy.parseURL(URLHandler
sStreamHandlerProxy.java:281)

     [java] at java.net.URL.<init>(URL.java:596)

     [java] at java.net.URL.<init>(URL.java:464)

     [java] at java.net.URL.<init>(URL.java:413)

     [java] at javax.crypto.SunJCE_c.b(DashoA13*..)

     [java] at javax.crypto.SunJCE_c.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_c.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_b.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_h.a(DashoA13*..)

     [java] at javax.crypto.SunJCE_h.a(DashoA13*..)

     [java] at javax.crypto.Cipher.c(DashoA13*..)

     [java] at javax.crypto.Cipher.b(DashoA13*..)

     [java] at javax.crypto.Cipher.getInstance(DashoA13*..)

     [java] at gnu.java.zrtp.ZRtp.<init>(ZRtp.java:329)

     [java] at
net.java.sip.communicator.impl.media.transform.zrtp.ZRTPTransformEngine.init
ialize(ZRTPTransformEngine.java:447)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.initializeRtpManager(Ca
llSessionImpl.java:2123)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.allocateMediaPorts(Call
SessionImpl.java:2026)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.createSessionDescriptio
n(CallSessionImpl.java:1419)

     [java] at
net.java.sip.communicator.impl.media.CallSessionImpl.createSdpOffer(CallSess
ionImpl.java:646)

     [java] at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImp
l.createOutgoingCall(OperationSetBasicTelephonySipImpl.java:298)

     [java] at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImp
l.createCall(OperationSetBasicTelephonySipImpl.java:165)

     [java] at
net.java.sip.communicator.impl.gui.main.call.CallManager$CreateCallThread.ru
n(CallManager.java:377)

I don’t know what it is this problem, and what is more yesterday I didn’t
get this problem doing the same call. Can anybody help me, please?

Another thing else related to zrtp, How could I deactivate the zrtp
encryption? I’m working with a server which catch the sip invite message
and it don’t tolerate the following attribute related to zrtp.

Thanks a lot, and best regards

Pedro Antonio Moreno Sanchez
Telecommunication Engineer & Researcher
Bioengineering and Telemedicine Group
ETSI Telecomunicación - UPM B.302.
Ciudad Universitaria s/n
28040 MADRID - SPAIN

Tel. +34 91 549 57 00 ext. 3407
Fax + 34 91 336 68 28
e-mail: <mailto:pmoreno@gbt.tfo.upm.es> pmoreno@gbt.tfo.upm.es

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#3

Pedro,

see inline please

Regards,
Werner

Pedro Antonio Moreno Sanchez schrieb:

I don't know what it is this problem, and what is more yesterday I didn't
get this problem doing the same call. Can anybody help me, please?

Did you update the Java version? If yes then the JCE security policy files
were installed again. To overcome this problem we modified the security stuff
to use another well known crypto library (BouncyCastle) instead of Sun JCE.

Another thing else related to zrtp, How could I deactivate the zrtp
encryption? I'm working with a server which catch the sip invite message
and it don't tolerate the following attribute related to zrtp.

Go to your SIP account wizard, select the advanced tab. On the advanced page
of the account wizrad you find a checkbox to enable/disable secure calls.

···

Thanks a lot, and best regards

Pedro Antonio Moreno Sanchez
Telecommunication Engineer & Researcher
Bioengineering and Telemedicine Group
ETSI Telecomunicaci�n - UPM B.302.
Ciudad Universitaria s/n
28040 MADRID - SPAIN

Tel. +34 91 549 57 00 ext. 3407
Fax + 34 91 336 68 28
e-mail: <mailto:pmoreno@gbt.tfo.upm.es> pmoreno@gbt.tfo.upm.es

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#4

All (mainly Yana and Lubo :wink: ),

to overcome the problem that Pedro explains below about the
restrictive SIP gateway/server that catches and filters unknown
SDP attributes lead me to some toughts:

- we need to implement another configuration (expert configuration)
  option that enables/disables the inclusion of the ZRTP SDP attribute.
  The usage of the SDP ZRTP attribute is optional according to the
  ZRTP draft. Question: do you have an idea where to place such a
  configuration checkbox? IMHO in the advanced tab of SIP account
  wizard?

Regards,
Werner

PS: hopefully the RTP stream isn't filtered :slight_smile: .

Werner Dittmann schrieb:

···

Another thing else related to zrtp, How could I deactivate the zrtp
encryption? I’m working with a server which catch the sip invite message
and it don’t tolerate the following attribute related to zrtp.

Thanks a lot, and best regards

Pedro Antonio Moreno Sanchez
Telecommunication Engineer & Researcher
Bioengineering and Telemedicine Group
ETSI Telecomunicación - UPM B.302.
Ciudad Universitaria s/n
28040 MADRID - SPAIN

Tel. +34 91 549 57 00 ext. 3407
Fax + 34 91 336 68 28
e-mail: <mailto:pmoreno@gbt.tfo.upm.es> pmoreno@gbt.tfo.upm.es

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#5

Hello Werner,

I definitely consider Yana and Emil more knowledgeable on the subject
but, since you explicitly listed me, I'm sharing my opinion here that
I agree with you on placing the checkbox on the Advanced tab of the
SIP account wizard. I guess it should be position in proximity to the
"Enable support to encrypt calls" checkbox. If the ZRTP SDP attribute
is only sent when the "Enable support to encrypt calls" checbox is
checked, I think the new checkbox should be disabled when the "Enable
support to encrypt calls" checkbox is unchecked.

Regards,
Lubo

···

On Wed, Apr 22, 2009 at 7:21 PM, Werner Dittmann <Werner.Dittmann@t-online.de> wrote:

All (mainly Yana and Lubo :wink: ),

to overcome the problem that Pedro explains below about the
restrictive SIP gateway/server that catches and filters unknown
SDP attributes lead me to some toughts:

- we need to implement another configuration (expert configuration)
option that enables/disables the inclusion of the ZRTP SDP attribute.
The usage of the SDP ZRTP attribute is optional according to the
ZRTP draft. Question: do you have an idea where to place such a
configuration checkbox? IMHO in the advanced tab of SIP account
wizard?

Regards,
Werner

PS: hopefully the RTP stream isn't filtered :slight_smile: .

Werner Dittmann schrieb:

Another thing else related to zrtp, How could I deactivate the zrtp
encryption? I’m working with a server which catch the sip invite message
and it don’t tolerate the following attribute related to zrtp.

Thanks a lot, and best regards

Pedro Antonio Moreno Sanchez
Telecommunication Engineer & Researcher
Bioengineering and Telemedicine Group
ETSI Telecomunicación - UPM B.302.
Ciudad Universitaria s/n
28040 MADRID - SPAIN

Tel. +34 91 549 57 00 ext. 3407
Fax + 34 91 336 68 28
e-mail: <mailto:pmoreno@gbt.tfo.upm.es> pmoreno@gbt.tfo.upm.es

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