[jitsi-users] Not able to call phones via SIP account


#1

This is my first attempt at using Jitsi for placing phone calls.

The SIP account that I registered in Jitsi (2.5) with proxy FQDN and all default settings came online right away.
Then I added a phone number as a contact and tried to call it. The call stays at Initiating call for a minute, then fails.

The same SIP account works fine in the ATA on the same network.

What am I missing?


#2

Why is Jitsi crashing when computer comes out of standby?
It runs Fedora 19 with KDE and every time it resumes from sleep Jitsi
is either no more running and ABRT has a crash report for JDK, or Jitsi
crashes soon after resume.
I tried removing open JDK and replacing it with the latest Oracle JDK
1.7-50 but nothing changed and Jitsi still crashing.

Does Jitsi require a specific JDK or JRE?

···

On Sat, 22 Feb 2014 10:15:14 -0500, <dubrovic@mail.md> wrote:

This is my first attempt at using Jitsi for placing phone calls.

The SIP account that I registered in Jitsi (2.5) with proxy FQDN and
all default settings came online right away.
Then I added a phone number as a contact and tried to call it. The
call stays at Initiating call for a minute, then fails.

The same SIP account works fine in the ATA on the same network.

What am I missing?

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#3

Just following up to make sure this question did not slip thru the
cracks.

···

On Sat, 22 Feb 2014 10:15:14 -0500, <dubrovic@mail.md> wrote:

This is my first attempt at using Jitsi for placing phone calls.

The SIP account that I registered in Jitsi (2.5) with proxy FQDN and
all default settings came online right away.
Then I added a phone number as a contact and tried to call it. The
call stays at Initiating call for a minute, then fails.

The same SIP account works fine in the ATA on the same network.

What am I missing?

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#4

Hi,

did you try: https://jitsi.org/faq#toc16

Regards
damencho

···

On Sat, Feb 22, 2014 at 5:15 PM, <dubrovic@mail.md> wrote:

This is my first attempt at using Jitsi for placing phone calls.

The SIP account that I registered in Jitsi (2.5) with proxy FQDN and all
default settings came online right away.
Then I added a phone number as a contact and tried to call it. The call
stays at Initiating call for a minute, then fails.

The same SIP account works fine in the ATA on the same network.

What am I missing?

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#5

Yes, and that did not work. The provider allows G722 and that was the
only codec I enabled.
The call failed right away, it did not connect with the error "Not
acceptable here".

···

On Mon, 24 Feb 2014 16:50:44 +0200, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

did you try: https://jitsi.org/faq#toc16

Regards
damencho

On Sat, Feb 22, 2014 at 5:15 PM, <dubrovic@mail.md> wrote:

This is my first attempt at using Jitsi for placing phone calls.

The SIP account that I registered in Jitsi (2.5) with proxy FQDN and all
default settings came online right away.
Then I added a phone number as a contact and tried to call it. The call
stays at Initiating call for a minute, then fails.

The same SIP account works fine in the ATA on the same network.

What am I missing?

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#6

So it seems is missing a codec, try enabling one by one ulaw, alaw ...
Receiving an error is ok, while staying to initiating ... or
connecting ... means there is a problem with communication most
probably fragmentation and requests or responses are not received.


#7

Enabling the other codecs seems kind of pointless as G722 is the only
one that provider supports.

As to fragmentation, what would affect our ATA that works over UDP too.
What am I missing?

···

On Mon, 24 Feb 2014 17:22:57 +0200, Damian Minkov <damencho@jitsi.org> wrote:

So it seems is missing a codec, try enabling one by one ulaw, alaw ...
Receiving an error is ok, while staying to initiating ... or
connecting ... means there is a problem with communication most
probably fragmentation and requests or responses are not received.

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#8

The difference is that Jitsi supports many more codecs (audio and
video) and features and when adding all of those to a udp packet may
exceed the frame size.
If you will not use video with this provider try disabling the video.

···

On Mon, Feb 24, 2014 at 5:39 PM, <dubrovic@mail.md> wrote:

Enabling the other codecs seems kind of pointless as G722 is the only
one that provider supports.

As to fragmentation, what would affect our ATA that works over UDP too.
What am I missing?

On Mon, 24 Feb 2014 17:22:57 +0200, Damian Minkov <damencho@jitsi.org> > wrote:

So it seems is missing a codec, try enabling one by one ulaw, alaw ...
Receiving an error is ok, while staying to initiating ... or
connecting ... means there is a problem with communication most
probably fragmentation and requests or responses are not received.

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#9

Hi,

Sorry about the confusion - I just spoke with the provider and they
advised that they only supported

G.711U G.729a gsm

Not G722! I asked them to enable GSM in our profile, enabled GSM in
Jitsi and disabled H.264 that was the only enabled video codec.

Still the same issue: call fails with the same "Not acceptable here"
error message.

···

On Mon, 24 Feb 2014 17:45:56 +0200, Damian Minkov <damencho@jitsi.org> wrote:

The difference is that Jitsi supports many more codecs (audio and
video) and features and when adding all of those to a udp packet may
exceed the frame size.
If you will not use video with this provider try disabling the video.

On Mon, Feb 24, 2014 at 5:39 PM, <dubrovic@mail.md> wrote:

Enabling the other codecs seems kind of pointless as G722 is the only
one that provider supports.

As to fragmentation, what would affect our ATA that works over UDP too.
What am I missing?

On Mon, 24 Feb 2014 17:22:57 +0200, Damian Minkov <damencho@jitsi.org> >> wrote:

So it seems is missing a codec, try enabling one by one ulaw, alaw ...
Receiving an error is ok, while staying to initiating ... or
connecting ... means there is a problem with communication most
probably fragmentation and requests or responses are not received.


#10

Did you tried with G.711U only? Gsm is not a good choice right now.

Providing the logs will also help.


#11

There is no G711u in Jitsi audio encodings.

···

On Mon, 24 Feb 2014 18:36:21 +0200, Damian Minkov <damencho@jitsi.org> wrote:

Did you tried with G.711U only? Gsm is not a good choice right now.

Providing the logs will also help.

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#12

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

···

On Mon, Feb 24, 2014 at 11:48 AM, <dubrovic@mail.md> wrote:

There is no G711u in Jitsi audio encodings.

On Mon, 24 Feb 2014 18:36:21 +0200, Damian Minkov <damencho@jitsi.org> > wrote:

Did you tried with G.711U only? Gsm is not a good choice right now.

Providing the logs will also help.

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#13

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

···

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.


#14

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

···

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven > <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

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users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
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#15

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

···

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.


#16

the echo cancellation for the other party is done on your end in
software for both skype and and jitsi.
If you are hearing echo then it is usually the other parties phone.
The latency is another issue lets solve one at a time.
Who is hearing the echo you or the other party?

Sincerely,

Russell Treleaven

···

On Mon, Feb 24, 2014 at 1:25 PM, <dubrovic@mail.md> wrote:

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven > <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >>> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

_______________________________________________
users mailing list
users@jitsi.org
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#17

I am hearing echo.

···

On Mon, 24 Feb 2014 13:32:28 -0500, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

the echo cancellation for the other party is done on your end in
software for both skype and and jitsi.
If you are hearing echo then it is usually the other parties phone.
The latency is another issue lets solve one at a time.
Who is hearing the echo you or the other party?

Sincerely,

Russell Treleaven

On Mon, Feb 24, 2014 at 1:25 PM, <dubrovic@mail.md> wrote:

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven >> <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >>>> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

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users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
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#18

call this sip uri sip:thetestcall@getonsip.com
select 2 to record your voice.
If you hear your own echo before you press # then the echo problem is
on your end.

let us know how it goes.

···

On Mon, Feb 24, 2014 at 5:57 PM, <dubrovic@mail.md> wrote:

I am hearing echo.

On Mon, 24 Feb 2014 13:32:28 -0500, Russell Treleaven > <rtreleaven@bunnykick.ca> wrote:

the echo cancellation for the other party is done on your end in
software for both skype and and jitsi.
If you are hearing echo then it is usually the other parties phone.
The latency is another issue lets solve one at a time.
Who is hearing the echo you or the other party?

Sincerely,

Russell Treleaven

On Mon, Feb 24, 2014 at 1:25 PM, <dubrovic@mail.md> wrote:

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven >>> <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >>>>> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

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#19

"Call failed - Not found"

···

On Mon, 24 Feb 2014 18:08:02 -0500, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

call this sip uri sip:thetestcall@getonsip.com
select 2 to record your voice.
If you hear your own echo before you press # then the echo problem is
on your end.

let us know how it goes.

On Mon, Feb 24, 2014 at 5:57 PM, <dubrovic@mail.md> wrote:

I am hearing echo.

On Mon, 24 Feb 2014 13:32:28 -0500, Russell Treleaven >> <rtreleaven@bunnykick.ca> wrote:

the echo cancellation for the other party is done on your end in
software for both skype and and jitsi.
If you are hearing echo then it is usually the other parties phone.
The latency is another issue lets solve one at a time.
Who is hearing the echo you or the other party?

Sincerely,

Russell Treleaven

On Mon, Feb 24, 2014 at 1:25 PM, <dubrovic@mail.md> wrote:

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven >>>> <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >>>>>> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
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#20

Sorry about that.
Jitsi sent the call to your itsp.
Maybe your itsp has a test number.
If not email me directly and I will give you a temporary test account to try.

mailto:rtreleaven@bunnykick.ca

···

On Mon, Feb 24, 2014 at 6:50 PM, <dubrovic@mail.md> wrote:

"Call failed - Not found"

On Mon, 24 Feb 2014 18:08:02 -0500, Russell Treleaven > <rtreleaven@bunnykick.ca> wrote:

call this sip uri sip:thetestcall@getonsip.com
select 2 to record your voice.
If you hear your own echo before you press # then the echo problem is
on your end.

let us know how it goes.

On Mon, Feb 24, 2014 at 5:57 PM, <dubrovic@mail.md> wrote:

I am hearing echo.

On Mon, 24 Feb 2014 13:32:28 -0500, Russell Treleaven >>> <rtreleaven@bunnykick.ca> wrote:

the echo cancellation for the other party is done on your end in
software for both skype and and jitsi.
If you are hearing echo then it is usually the other parties phone.
The latency is another issue lets solve one at a time.
Who is hearing the echo you or the other party?

Sincerely,

Russell Treleaven

On Mon, Feb 24, 2014 at 1:25 PM, <dubrovic@mail.md> wrote:

The other party is using a regular phone. So this is my echo
cancellation, which makes sense as under Linux the Realtek driver has no
such option whatsoever. Would you know why Skype is not affected by echo
and latency though?

On Mon, 24 Feb 2014 12:49:20 -0500, Russell Treleaven >>>>> <rtreleaven@bunnykick.ca> wrote:

If the other party is hearing echo it is most likely your echo
canceller having issues.
If you are hearing echo than its the most likely the other person's
echo canceller.
http://www.voip-info.org/wiki/view/Causes+of+Echo

As far as delay goes.
I assume the data path should be similar between your ata to the itsp
as it is for your pc to the itsp so that should not be the problem.
My guess is that the jitter buffer is larger on Jitsi then on your pc.
I am new to Jitsi so I am not sure if that is configurable or not.

On Mon, Feb 24, 2014 at 12:38 PM, <dubrovic@mail.md> wrote:

PCMU works, thank you!

In the call there is lots of echo and significant latency between
saying
something and that being heard on the other side. There is of course
some echo at times when ATA is used, but nothing even close to the
amount of echo and latency as with Jitsi.

Can you recommend any measures to control echo and delay?

On Mon, 24 Feb 2014 11:52:55 -0500, Russell Treleaven >>>>>>> <rtreleaven@bunnykick.ca> wrote:

G711u is another way of saying PCMU.
Can you got a packet catpure using wireshark or similar?
post it to a pastebin and I have a look.

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
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