[jitsi-users] IM chat window doesn't open


#1

Hello,
Jitsi is working fine for me as a SIP client - softphone.
But when I configured an XMPP account and registered sucessfully, IM chat window can't open at all. Neither right click on contact and choosing "Send message" nor clicking chat button work.
System configuration:
Ubuntu 10.4
sun-java6-jre - version 6.30
Jitsi 1.0-beta1-nightly.build.3820 - older versions don't work as well

Record in sip-communicator0.log.0 :
09:21:27.369 SEVERE: util.UtilActivator.uncaughtException().88 An uncaught exception occurred in thread=Thread[AWT-EventQueue-0,6,
main] and message was: null
java.lang.NullPointerException
         at net.java.sip.communicator.util.swing.SIPCommFrame.setKeybindingInput(SIPCommFrame.java:236)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindow.<init>(ChatWindow.java:126)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getChatContainer(ChatWindowManager.java:910)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.createChat(ChatWindowManager.java:849)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:442)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:402)
         at net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:386)
         at net.java.sip.communicator.impl.gui.DefaultContactEventHandler$RunMessageWindow.run(DefaultContactEventHandler.java:96)
         at java.awt.event.InvocationEvent.dispatch(InvocationEvent.java:209)
         at java.awt.EventQueue.dispatchEventImpl(EventQueue.java:646)
         at java.awt.EventQueue.access$000(EventQueue.java:84)
         at java.awt.EventQueue$1.run(EventQueue.java:607)
         at java.awt.EventQueue$1.run(EventQueue.java:605)
         at java.security.AccessController.doPrivileged(Native Method)
         at java.security.AccessControlContext$1.doIntersectionPrivilege(AccessControlContext.java:87)
         at java.awt.EventQueue.dispatchEvent(EventQueue.java:616)
         at java.awt.EventDispatchThread.pumpOneEventForFilters(EventDispatchThread.java:269)
         at java.awt.EventDispatchThread.pumpEventsForFilter(EventDispatchThread.java:184)
         at java.awt.EventDispatchThread.pumpEventsForHierarchy(EventDispatchThread.java:174)
         at java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:169)
         at java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:161)
         at java.awt.EventDispatchThread.run(EventDispatchThread.java:122)

My workaround - usage of sun-java6-jre - version 6.26 solved the problem, but other important functionalities stopped working then (e.g. SSL VPN connection).

Please, could You tune Jitsi to work with sun-java6-jre - version 6.30 ?

Thank You in advance
Best Regards
Petr Valik


#2

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the closest
candidates that I can find. The limitations I see are that jitsi needs the
ability to select wider bandwidth (audio and video) codecs. These codecs are
probably too bandwidth intensive for most applications, but they are essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a particular codec,
like SILK-V3 or RAW-PCM? (Again, I'm not a coder or developer, so the question
is probably naive.) An application that uses very wideband audio codecs is
Navigator from audiocompass.com, but they don't have and are never going to
include video. There is a great demand among Internet broadcasters for a Skype
replacement, but we're just not there yet. jitsi seems like an excellent
candidate to make the leap.

MCG


#3

HI,

there seems to be some bugs with keybinding plugin in that version
which prevents normal running with that java version. Those problem
seems to be fixed in latest versions, at least I had no problem
running it on ubuntu Lucid with java 1.6.30.
You can try installing latest nightly and try with it.
http://download.jitsi.org/jitsi/nightly/debian/

Regards
damencho

···

On Thu, Jan 26, 2012 at 12:04 PM, Valík Petr <petr.valik@tieto.com> wrote:

Hello,
Jitsi is working fine for me as a SIP client - softphone.
But when I configured an XMPP account and registered sucessfully, IM chat
window can't open at all. Neither right click on contact and choosing "Send
message" nor clicking chat button work.
System configuration:
Ubuntu 10.4
sun-java6-jre - version 6.30
Jitsi 1.0-beta1-nightly.build.3820 - older versions don't work as well

Record in sip-communicator0.log.0 :
09:21:27.369 SEVERE: util.UtilActivator.uncaughtException().88 An uncaught
exception occurred in thread=Thread[AWT-EventQueue-0,6,
main] and message was: null
java.lang.NullPointerException
at
net.java.sip.communicator.util.swing.SIPCommFrame.setKeybindingInput(SIPCommFrame.java:236)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindow.<init>(ChatWindow.java:126)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getChatContainer(ChatWindowManager.java:910)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.createChat(ChatWindowManager.java:849)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:442)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:402)
at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:386)
at
net.java.sip.communicator.impl.gui.DefaultContactEventHandler$RunMessageWindow.run(DefaultContactEventHandler.java:96)
at java.awt.event.InvocationEvent.dispatch(InvocationEvent.java:209)
at java.awt.EventQueue.dispatchEventImpl(EventQueue.java:646)
at java.awt.EventQueue.access$000(EventQueue.java:84)
at java.awt.EventQueue$1.run(EventQueue.java:607)
at java.awt.EventQueue$1.run(EventQueue.java:605)
at java.security.AccessController.doPrivileged(Native Method)
at
java.security.AccessControlContext$1.doIntersectionPrivilege(AccessControlContext.java:87)
at java.awt.EventQueue.dispatchEvent(EventQueue.java:616)
at
java.awt.EventDispatchThread.pumpOneEventForFilters(EventDispatchThread.java:269)
at
java.awt.EventDispatchThread.pumpEventsForFilter(EventDispatchThread.java:184)
at
java.awt.EventDispatchThread.pumpEventsForHierarchy(EventDispatchThread.java:174)
at
java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:169)
at
java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:161)
at java.awt.EventDispatchThread.run(EventDispatchThread.java:122)

My workaround - usage of sun-java6-jre - version 6.26 solved the problem,
but other important functionalities stopped working then (e.g. SSL VPN
connection).

Please, could You tune Jitsi to work with sun-java6-jre - version 6.30 ?

Thank You in advance
Best Regards
Petr Valik


#4

Hey there,

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the closest
candidates that I can find. The limitations I see are that jitsi needs the
ability to select wider bandwidth (audio and video) codecs. These codecs are
probably too bandwidth intensive for most applications, but they are essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a particular codec,
like SILK-V3 or RAW-PCM?

We currently support Silk in 24KHz and Speex in 32 KHz. Are these good
enough?

Adding support for RAW PCM in 44KHz wouldn't be that hard to implement
but it would probably still require a couple of days or so. The result
would of course be quite bandwidth consuming.

Cheers,
Emil

···

On 26.01.12 17:39, Mediacast Guy wrote:

(Again, I'm not a coder or developer, so the question
is probably naive.) An application that uses very wideband audio codecs is
Navigator from audiocompass.com, but they don't have and are never going to
include video. There is a great demand among Internet broadcasters for a Skype
replacement, but we're just not there yet. jitsi seems like an excellent
candidate to make the leap.

MCG

--
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


#5

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 6:39:20 PM
Subject: [jitsi-users] Codec Selection

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very
good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in
SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the
closest
candidates that I can find. The limitations I see are that jitsi
needs the
ability to select wider bandwidth (audio and video) codecs. These
codecs are
probably too bandwidth intensive for most applications, but they are
essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a
particular codec,
like SILK-V3 or RAW-PCM?

You can select the used codecs in the options -> audio -> encodings menu. Very easy, just tick what you want and untick what you dont. The ordering is relevant too - the higher the codec is on the list, the higher its priority is (set the lower quality codecs lower to have smaller chance of being elected).
I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz (if the other side uses other software, most have speex but not many silk), then for fallback PCMU (i think this is the raw pcm @ 8 KHz) for compatibility with virtually every voip client out there (the trixbox server we use for SIP accepts only PCMU/PCMA).
The wideband codecs Silk and Speex, despite having wider audio range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has a 10 KB/s steady stream, Silk and Speex have variable bitrate, around 5-10 KB/s, mostly around 7 KB/s. They work fine even over broadband.
Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low bandwidth and has quite good picture quality, whereas h263 uses more bandwidth and has poorer quality.

From my experience Jitsi works well over Jabber-based networks (i use Google Talk) and SIP (Asterix/trixbox server)

.
(Again, I'm not a coder or developer, so the

···

----- Original Message -----

question
is probably naive.) An application that uses very wideband audio
codecs is
Navigator from audiocompass.com, but they don't have and are never
going to
include video. There is a great demand among Internet broadcasters
for a Skype
replacement, but we're just not there yet. jitsi seems like an
excellent
candidate to make the leap.

MCG

--
O zi buna,

Kertesz Laszlo


#6

Hi,
Thanks a lot - it works fine with Jitsi version 1.0-beta1-nightly.build.3878 <http://download.jitsi.org/jitsi/nightly/debian/jitsi_1.0-beta1-nightly.build.3878_i386.deb>
and Java 1.6.30

Best Regards
Petr V.

···

On 01/26/2012 11:46 AM, Damian Minkov wrote:

HI,

there seems to be some bugs with keybinding plugin in that version
which prevents normal running with that java version. Those problem
seems to be fixed in latest versions, at least I had no problem
running it on ubuntu Lucid with java 1.6.30.
You can try installing latest nightly and try with it.
http://download.jitsi.org/jitsi/nightly/debian/

Regards
damencho

On Thu, Jan 26, 2012 at 12:04 PM, Val�k Petr<petr.valik@tieto.com> wrote:

Hello,
Jitsi is working fine for me as a SIP client - softphone.
But when I configured an XMPP account and registered sucessfully, IM chat
window can't open at all. Neither right click on contact and choosing "Send
message" nor clicking chat button work.
System configuration:
Ubuntu 10.4
sun-java6-jre - version 6.30
Jitsi 1.0-beta1-nightly.build.3820 - older versions don't work as well

Record in sip-communicator0.log.0 :
09:21:27.369 SEVERE: util.UtilActivator.uncaughtException().88 An uncaught
exception occurred in thread=Thread[AWT-EventQueue-0,6,
main] and message was: null
java.lang.NullPointerException
        at
net.java.sip.communicator.util.swing.SIPCommFrame.setKeybindingInput(SIPCommFrame.java:236)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindow.<init>(ChatWindow.java:126)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getChatContainer(ChatWindowManager.java:910)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.createChat(ChatWindowManager.java:849)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:442)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:402)
        at
net.java.sip.communicator.impl.gui.main.chat.ChatWindowManager.getContactChat(ChatWindowManager.java:386)
        at
net.java.sip.communicator.impl.gui.DefaultContactEventHandler$RunMessageWindow.run(DefaultContactEventHandler.java:96)
        at java.awt.event.InvocationEvent.dispatch(InvocationEvent.java:209)
        at java.awt.EventQueue.dispatchEventImpl(EventQueue.java:646)
        at java.awt.EventQueue.access$000(EventQueue.java:84)
        at java.awt.EventQueue$1.run(EventQueue.java:607)
        at java.awt.EventQueue$1.run(EventQueue.java:605)
        at java.security.AccessController.doPrivileged(Native Method)
        at
java.security.AccessControlContext$1.doIntersectionPrivilege(AccessControlContext.java:87)
        at java.awt.EventQueue.dispatchEvent(EventQueue.java:616)
        at
java.awt.EventDispatchThread.pumpOneEventForFilters(EventDispatchThread.java:269)
        at
java.awt.EventDispatchThread.pumpEventsForFilter(EventDispatchThread.java:184)
        at
java.awt.EventDispatchThread.pumpEventsForHierarchy(EventDispatchThread.java:174)
        at
java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:169)
        at
java.awt.EventDispatchThread.pumpEvents(EventDispatchThread.java:161)
        at java.awt.EventDispatchThread.run(EventDispatchThread.java:122)

My workaround - usage of sun-java6-jre - version 6.26 solved the problem,
but other important functionalities stopped working then (e.g. SSL VPN
connection).

Please, could You tune Jitsi to work with sun-java6-jre - version 6.30 ?

Thank You in advance
Best Regards
Petr Valik


#7

You've now inspired me to do some additional testing. With our first tests, the
audio was not as good as Skype. Is there a way to FORCE jitsi to use SILK or
Speex? In retrospect, it is possible that we were using a lesser codec, like
g.722, because of my lack of familiarity with jitsi.

Also, what about HD video, like H.264? For the record, we tested H.263 and H.264
with Bria, and the video quality was not as good as Skype. In that case, though,
there was likely a problem on the remote computer. We're addressing that problem
before we test again.

You are correct that the applications are bandwidth consuming. That's probably
why no one is working to include these extraordinary quality codecs. We have 5Mb
uploads. With time, we hope to get ever better speeds. The quality requirement
is worth the bandwidth.

I'm getting excited about jitsi's potential for this specialized application.
Good job.

MCG

···

On 1/26/2012 11:57 AM, Emil Ivov wrote:

Hey there,

On 26.01.12 17:39, Mediacast Guy wrote:

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the closest
candidates that I can find. The limitations I see are that jitsi needs the
ability to select wider bandwidth (audio and video) codecs. These codecs are
probably too bandwidth intensive for most applications, but they are essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a particular codec,
like SILK-V3 or RAW-PCM?

We currently support Silk in 24KHz and Speex in 32 KHz. Are these good
enough?

Adding support for RAW PCM in 44KHz wouldn't be that hard to implement
but it would probably still require a couple of days or so. The result
would of course be quite bandwidth consuming.


#8

Great information. Thanks. By the way, how do you use it with Google Talk? We've
only used SIP.

MCG

···

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:

----- Original Message -----

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 6:39:20 PM
Subject: [jitsi-users] Codec Selection

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very
good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in
SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the
closest
candidates that I can find. The limitations I see are that jitsi
needs the
ability to select wider bandwidth (audio and video) codecs. These
codecs are
probably too bandwidth intensive for most applications, but they are
essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a
particular codec,
like SILK-V3 or RAW-PCM?

You can select the used codecs in the options -> audio -> encodings menu. Very easy, just tick what you want and untick what you dont. The ordering is relevant too - the higher the codec is on the list, the higher its priority is (set the lower quality codecs lower to have smaller chance of being elected).
I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz (if the other side uses other software, most have speex but not many silk), then for fallback PCMU (i think this is the raw pcm @ 8 KHz) for compatibility with virtually every voip client out there (the trixbox server we use for SIP accepts only PCMU/PCMA).
The wideband codecs Silk and Speex, despite having wider audio range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has a 10 KB/s steady stream, Silk and Speex have variable bitrate, around 5-10 KB/s, mostly around 7 KB/s. They work fine even over broadband.
Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low bandwidth and has quite good picture quality, whereas h263 uses more bandwidth and has poorer quality.
From my experience Jitsi works well over Jabber-based networks (i use Google Talk) and SIP (Asterix/trixbox server)
.
(Again, I'm not a coder or developer, so the

question
is probably naive.) An application that uses very wideband audio
codecs is
Navigator from audiocompass.com, but they don't have and are never
going to
include video. There is a great demand among Internet broadcasters
for a Skype
replacement, but we're just not there yet. jitsi seems like an
excellent
candidate to make the leap.

MCG


#9

So I connected with my Google Talk account. When I dial my cell phone number
with jitsi, the cell phone rings, and I answer it. The call still shows as
"connecting" on jitsi. After a few seconds, jitsi hangs up. When I tried the
same procedure with my SIP account, everything connects and works as expected.
Am I missing something in configuring Google Talk?

MCG

···

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:

----- Original Message -----

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 6:39:20 PM
Subject: [jitsi-users] Codec Selection

Like many others, our Internet broadcasting group is trying to find a
replacement for Skype. Even though Skype audio and video are very
good, Skype's
other issues (renegotiation) are problematic. While I'm no expert in
SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the
closest
candidates that I can find. The limitations I see are that jitsi
needs the
ability to select wider bandwidth (audio and video) codecs. These
codecs are
probably too bandwidth intensive for most applications, but they are
essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a
particular codec,
like SILK-V3 or RAW-PCM?

You can select the used codecs in the options -> audio -> encodings menu. Very easy, just tick what you want and untick what you dont. The ordering is relevant too - the higher the codec is on the list, the higher its priority is (set the lower quality codecs lower to have smaller chance of being elected).
I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz (if the other side uses other software, most have speex but not many silk), then for fallback PCMU (i think this is the raw pcm @ 8 KHz) for compatibility with virtually every voip client out there (the trixbox server we use for SIP accepts only PCMU/PCMA).
The wideband codecs Silk and Speex, despite having wider audio range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has a 10 KB/s steady stream, Silk and Speex have variable bitrate, around 5-10 KB/s, mostly around 7 KB/s. They work fine even over broadband.
Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low bandwidth and has quite good picture quality, whereas h263 uses more bandwidth and has poorer quality.
From my experience Jitsi works well over Jabber-based networks (i use Google Talk) and SIP (Asterix/trixbox server)
.
(Again, I'm not a coder or developer, so the

question
is probably naive.) An application that uses very wideband audio
codecs is
Navigator from audiocompass.com, but they don't have and are never
going to
include video. There is a great demand among Internet broadcasters
for a Skype
replacement, but we're just not there yet. jitsi seems like an
excellent
candidate to make the leap.

MCG


#10

My experience is that to call a cellphone or landline with GoogleTalk
you have to have GoogleVoice. Then you put the number you are calling
in the window above the contact list that says "Enter name or number"
When you click on the green telephone icon that appears by the number
you have entered, select the GoogleTalk option. It works I add,
however, that the quality has not been as good as, for example, making
the call directly over GoogleTalk without going through Jitsi.

I have not had success in calling out on GoogleTalk to do voice chat
with another GoogleTalk account. I get this error message. "Could not
establish connection. ICE failed" HOwever, if the other party calls
me, I hear fine.

Paul

···

El jue, 26-01-2012 a las 14:24 -0500, Mediacast Guy escribió:

So I connected with my Google Talk account. When I dial my cell phone number
with jitsi, the cell phone rings, and I answer it. The call still shows as
"connecting" on jitsi. After a few seconds, jitsi hangs up. When I tried the
same procedure with my SIP account, everything connects and works as expected.
Am I missing something in configuring Google Talk?

MCG

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:
>
> ----- Original Message -----
>> From: "Mediacast Guy" <mediacastguy@gmail.com>
>> To: users@jitsi.java.net
>> Sent: Thursday, January 26, 2012 6:39:20 PM
>> Subject: [jitsi-users] Codec Selection
>>
>> Like many others, our Internet broadcasting group is trying to find a
>> replacement for Skype. Even though Skype audio and video are very
>> good, Skype's
>> other issues (renegotiation) are problematic. While I'm no expert in
>> SIP, jitsi,
>> or coding, I have tested jitsi, and it appears to be one of the
>> closest
>> candidates that I can find. The limitations I see are that jitsi
>> needs the
>> ability to select wider bandwidth (audio and video) codecs. These
>> codecs are
>> probably too bandwidth intensive for most applications, but they are
>> essential
>> for broadcasting.
>>
>> My question is, how difficult is it to code jitsi to accept a
>> particular codec,
>> like SILK-V3 or RAW-PCM?
> You can select the used codecs in the options -> audio -> encodings menu. Very easy, just tick what you want and untick what you dont. The ordering is relevant too - the higher the codec is on the list, the higher its priority is (set the lower quality codecs lower to have smaller chance of being elected).
> I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz (if the other side uses other software, most have speex but not many silk), then for fallback PCMU (i think this is the raw pcm @ 8 KHz) for compatibility with virtually every voip client out there (the trixbox server we use for SIP accepts only PCMU/PCMA).
> The wideband codecs Silk and Speex, despite having wider audio range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has a 10 KB/s steady stream, Silk and Speex have variable bitrate, around 5-10 KB/s, mostly around 7 KB/s. They work fine even over broadband.
> Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low bandwidth and has quite good picture quality, whereas h263 uses more bandwidth and has poorer quality.
> From my experience Jitsi works well over Jabber-based networks (i use Google Talk) and SIP (Asterix/trixbox server)
> .
> (Again, I'm not a coder or developer, so the
>> question
>> is probably naive.) An application that uses very wideband audio
>> codecs is
>> Navigator from audiocompass.com, but they don't have and are never
>> going to
>> include video. There is a great demand among Internet broadcasters
>> for a Skype
>> replacement, but we're just not there yet. jitsi seems like an
>> excellent
>> candidate to make the leap.
>>
>> MCG
>>


#11

Folks

When I try to add a contact, I get message "network failure occurred".
When I click on "more info" it tells me the number is not a valid
string. What am I doing wrong? It did not use to happen but it has
begun to happen recently.

Also, I have not discovered how to edit a contact once created.

Thank you for your help

Paul


#12

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 7:20:48 PM
Subject: [jitsi-users] Re: Codec Selection

Great information. Thanks. By the way, how do you use it with Google
Talk? We've
only used SIP.

Just add a google talk account, options -> accounts -> add.

There is one thing to be aware if both sides use Jitsi with google talk or XMPP - Jitsi supports zrtp on-the-fly end-to-end encryption that is activated if both sides support it (for example both sides use Jitsi). The negociation period is a few seconds (extreme case i had 21) and until that point both sides are shown as connected, but audio usually doesnt work both ways (one side may hear the other). Only after the security is established (there is an audio signal + 4 character string is shown to both sides) the call becomes 2-way.

···

----- Original Message -----

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:
>
> ----- Original Message -----
>> From: "Mediacast Guy" <mediacastguy@gmail.com>
>> To: users@jitsi.java.net
>> Sent: Thursday, January 26, 2012 6:39:20 PM
>> Subject: [jitsi-users] Codec Selection
>>
>> Like many others, our Internet broadcasting group is trying to
>> find a
>> replacement for Skype. Even though Skype audio and video are very
>> good, Skype's
>> other issues (renegotiation) are problematic. While I'm no expert
>> in
>> SIP, jitsi,
>> or coding, I have tested jitsi, and it appears to be one of the
>> closest
>> candidates that I can find. The limitations I see are that jitsi
>> needs the
>> ability to select wider bandwidth (audio and video) codecs. These
>> codecs are
>> probably too bandwidth intensive for most applications, but they
>> are
>> essential
>> for broadcasting.
>>
>> My question is, how difficult is it to code jitsi to accept a
>> particular codec,
>> like SILK-V3 or RAW-PCM?
> You can select the used codecs in the options -> audio -> encodings
> menu. Very easy, just tick what you want and untick what you dont.
> The ordering is relevant too - the higher the codec is on the
> list, the higher its priority is (set the lower quality codecs
> lower to have smaller chance of being elected).
> I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz
> (if the other side uses other software, most have speex but not
> many silk), then for fallback PCMU (i think this is the raw pcm @
> 8 KHz) for compatibility with virtually every voip client out
> there (the trixbox server we use for SIP accepts only PCMU/PCMA).
> The wideband codecs Silk and Speex, despite having wider audio
> range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has
> a 10 KB/s steady stream, Silk and Speex have variable bitrate,
> around 5-10 KB/s, mostly around 7 KB/s. They work fine even over
> broadband.
> Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low
> bandwidth and has quite good picture quality, whereas h263 uses
> more bandwidth and has poorer quality.
> From my experience Jitsi works well over Jabber-based networks (i
> use Google Talk) and SIP (Asterix/trixbox server)
> .
> (Again, I'm not a coder or developer, so the
>> question
>> is probably naive.) An application that uses very wideband audio
>> codecs is
>> Navigator from audiocompass.com, but they don't have and are never
>> going to
>> include video. There is a great demand among Internet broadcasters
>> for a Skype
>> replacement, but we're just not there yet. jitsi seems like an
>> excellent
>> candidate to make the leap.
>>
>> MCG
>>

--
O zi buna,

Kertesz Laszlo


#13

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 9:24:00 PM
Subject: [jitsi-users] Re: Codec Selection

So I connected with my Google Talk account. When I dial my cell phone
number
with jitsi, the cell phone rings, and I answer it. The call still
shows as
"connecting" on jitsi. After a few seconds, jitsi hangs up. When I
tried the
same procedure with my SIP account, everything connects and works as
expected.
Am I missing something in configuring Google Talk?

Unfortunately i cannot help with call-out as i never used it with google talk.
But maybe somebody from the dev team might help with this.

There is a telephony section in the google talk account configuration panel. Maybe that has something to do with it?

···

----- Original Message -----

MCG

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:
>
> ----- Original Message -----
>> From: "Mediacast Guy" <mediacastguy@gmail.com>
>> To: users@jitsi.java.net
>> Sent: Thursday, January 26, 2012 6:39:20 PM
>> Subject: [jitsi-users] Codec Selection
>>
>> Like many others, our Internet broadcasting group is trying to
>> find a
>> replacement for Skype. Even though Skype audio and video are very
>> good, Skype's
>> other issues (renegotiation) are problematic. While I'm no expert
>> in
>> SIP, jitsi,
>> or coding, I have tested jitsi, and it appears to be one of the
>> closest
>> candidates that I can find. The limitations I see are that jitsi
>> needs the
>> ability to select wider bandwidth (audio and video) codecs. These
>> codecs are
>> probably too bandwidth intensive for most applications, but they
>> are
>> essential
>> for broadcasting.
>>
>> My question is, how difficult is it to code jitsi to accept a
>> particular codec,
>> like SILK-V3 or RAW-PCM?
> You can select the used codecs in the options -> audio -> encodings
> menu. Very easy, just tick what you want and untick what you dont.
> The ordering is relevant too - the higher the codec is on the
> list, the higher its priority is (set the lower quality codecs
> lower to have smaller chance of being elected).
> I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz
> (if the other side uses other software, most have speex but not
> many silk), then for fallback PCMU (i think this is the raw pcm @
> 8 KHz) for compatibility with virtually every voip client out
> there (the trixbox server we use for SIP accepts only PCMU/PCMA).
> The wideband codecs Silk and Speex, despite having wider audio
> range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has
> a 10 KB/s steady stream, Silk and Speex have variable bitrate,
> around 5-10 KB/s, mostly around 7 KB/s. They work fine even over
> broadband.
> Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low
> bandwidth and has quite good picture quality, whereas h263 uses
> more bandwidth and has poorer quality.
> From my experience Jitsi works well over Jabber-based networks (i
> use Google Talk) and SIP (Asterix/trixbox server)
> .
> (Again, I'm not a coder or developer, so the
>> question
>> is probably naive.) An application that uses very wideband audio
>> codecs is
>> Navigator from audiocompass.com, but they don't have and are never
>> going to
>> include video. There is a great demand among Internet broadcasters
>> for a Skype
>> replacement, but we're just not there yet. jitsi seems like an
>> excellent
>> candidate to make the leap.
>>
>> MCG
>>

--
O zi buna,

Kertesz Laszlo


#14

Hey Paul,

Could we have a look at your logs?

Thanks,
Emil

···

On 28.01.12 22:19, Carola y Pablo wrote:

Folks

When I try to add a contact, I get message "network failure occurred".
When I click on "more info" it tells me the number is not a valid
string. What am I doing wrong? It did not use to happen but it has
begun to happen recently.

Also, I have not discovered how to edit a contact once created.

Thank you for your help

Paul

--
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


#15

Also, how can you tell which codec jisti has negotiated with the other end? When
I first tried jitsi, I tried several at once. Once SIP client actually displays
the name of the codec in use on the dial page. That's a very handy feature.

MCG

···

On 1/26/2012 12:36 PM, Kertesz Laszlo wrote:

----- Original Message -----

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 7:20:48 PM
Subject: [jitsi-users] Re: Codec Selection

Great information. Thanks. By the way, how do you use it with Google
Talk? We've
only used SIP.

Just add a google talk account, options -> accounts -> add.

There is one thing to be aware if both sides use Jitsi with google talk or XMPP - Jitsi supports zrtp on-the-fly end-to-end encryption that is activated if both sides support it (for example both sides use Jitsi). The negociation period is a few seconds (extreme case i had 21) and until that point both sides are shown as connected, but audio usually doesnt work both ways (one side may hear the other). Only after the security is established (there is an audio signal + 4 character string is shown to both sides) the call becomes 2-way.

On 1/26/2012 12:10 PM, Kertesz Laszlo wrote:

----- Original Message -----

From: "Mediacast Guy" <mediacastguy@gmail.com>
To: users@jitsi.java.net
Sent: Thursday, January 26, 2012 6:39:20 PM
Subject: [jitsi-users] Codec Selection

Like many others, our Internet broadcasting group is trying to
find a
replacement for Skype. Even though Skype audio and video are very
good, Skype's
other issues (renegotiation) are problematic. While I'm no expert
in
SIP, jitsi,
or coding, I have tested jitsi, and it appears to be one of the
closest
candidates that I can find. The limitations I see are that jitsi
needs the
ability to select wider bandwidth (audio and video) codecs. These
codecs are
probably too bandwidth intensive for most applications, but they
are
essential
for broadcasting.

My question is, how difficult is it to code jitsi to accept a
particular codec,
like SILK-V3 or RAW-PCM?

You can select the used codecs in the options -> audio -> encodings
menu. Very easy, just tick what you want and untick what you dont.
The ordering is relevant too - the higher the codec is on the
list, the higher its priority is (set the lower quality codecs
lower to have smaller chance of being elected).
I use 3 codecs - silk 24 KHz is highest, then speex 32 and 16 KHz
(if the other side uses other software, most have speex but not
many silk), then for fallback PCMU (i think this is the raw pcm @
8 KHz) for compatibility with virtually every voip client out
there (the trixbox server we use for SIP accepts only PCMU/PCMA).
The wideband codecs Silk and Speex, despite having wider audio
range (24/32 KHz) use less bandwidth than PCMU ( 8 KHz). PCMU has
a 10 KB/s steady stream, Silk and Speex have variable bitrate,
around 5-10 KB/s, mostly around 7 KB/s. They work fine even over
broadband.
Jitsi uses 2 video codecs - h264 and h263-1998. h264 uses very low
bandwidth and has quite good picture quality, whereas h263 uses
more bandwidth and has poorer quality.
From my experience Jitsi works well over Jabber-based networks (i
use Google Talk) and SIP (Asterix/trixbox server)
.
(Again, I'm not a coder or developer, so the

question
is probably naive.) An application that uses very wideband audio
codecs is
Navigator from audiocompass.com, but they don't have and are never
going to
include video. There is a great demand among Internet broadcasters
for a Skype
replacement, but we're just not there yet. jitsi seems like an
excellent
candidate to make the leap.

MCG


#16

Emil,

About the difficulty adding contacts: Although the "network failure
occurred" message has ocurred several times, now it is working fine. So
I guess there is no need to send the log file at this time.

About editing contacts--with this I do need help. I do not see any way
to edit a contact once created--for example to correct a phone number,
or change the provider. I press the right mouse button on the contact
and I get a dropdown menu, but editing does not seem to be an option.
How does that work?

Also, I am using the "unstable" version nightly builds, because they
allow me to call using GoogleVoice. I see they don't update
automatically and I see there is an instruction to add a line to use the
repository. I am a novice in all this. Do I just paste that line in
the Terminal or are there other commands I must use? I am using Ubuntu
11.10

Thank you

Paul

···

El lun, 30-01-2012 a las 18:05 +0100, Emil Ivov escribió:

Hey Paul,

Could we have a look at your logs?

Thanks,
Emil

On 28.01.12 22:19, Carola y Pablo wrote:
> Folks
>
> When I try to add a contact, I get message "network failure occurred".
> When I click on "more info" it tells me the number is not a valid
> string. What am I doing wrong? It did not use to happen but it has
> begun to happen recently.
>
> Also, I have not discovered how to edit a contact once created.
>
> Thank you for your help
>
> Paul
>
>


#17

Hi,

Emil,

About the difficulty adding contacts: Although the "network failure
occurred" message has ocurred several times, now it is working fine. So
I guess there is no need to send the log file at this time.

About editing contacts--with this I do need help. I do not see any way
to edit a contact once created--for example to correct a phone number,
or change the provider. I press the right mouse button on the contact
and I get a dropdown menu, but editing does not seem to be an option.
How does that work?

There is no option to edit contact's address once they are created.
You can only edit its display name by renaming it. To change the
address recreate the contact.

Also, I am using the "unstable" version nightly builds, because they
allow me to call using GoogleVoice. I see they don't update
automatically and I see there is an instruction to add a line to use the
repository. I am a novice in all this. Do I just paste that line in
the Terminal or are there other commands I must use? I am using Ubuntu
11.10

Add new file /etc/apt/sources.list.d/jitsi-nightly.list
$sudo gedit /etc/apt/sources.list.d/jitsi-nightly.list
Adding in it the following line:
deb http://download.jitsi.org/nightly/deb unstable/

Cheers
damencho

···

On Tue, Jan 31, 2012 at 2:02 PM, Carola y Pablo <stuckybyler2@yahoo.es> wrote:

Thank you

Paul

El lun, 30-01-2012 a las 18:05 +0100, Emil Ivov escribió:

Hey Paul,

Could we have a look at your logs?

Thanks,
Emil

On 28.01.12 22:19, Carola y Pablo wrote:
> Folks
>
> When I try to add a contact, I get message "network failure occurred".
> When I click on "more info" it tells me the number is not a valid
> string. What am I doing wrong? It did not use to happen but it has
> begun to happen recently.
>
> Also, I have not discovered how to edit a contact once created.
>
> Thank you for your help
>
> Paul
>
>