Marek Paśnikowski | 1 Jan 2012 22:22
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Can't confirm registration

I have registered for a new sip.linphone.org account. However, in the 
meantime I accidentaly purged contents of my mailbox. Now I am stuck. I do 
want my selected username, that's why I don't want to register a new 
username. Please, resend the confirmation mail or delete the application from 
queue, so I can register again.
alekcejk | 1 Jan 2012 23:08

Sound disappears if "Enable echo cancellation" disabled

Hi,

I have a problem with sound if "Enable echo cancellation" option disabled.
In some calls sound disappears when this option disabled but volume
indicators shows sound volume as if sound was fine.
Sound appears after Pause/Resume pressed.
There are no any unusual messages in debug window.
No such problem if "Enable echo cancellation" option is enabled.

I also noticed that after I press button that disables microphone
and then press Pause/Resume then button looks as if microphone
was not disabled but actually it is disabled.

--

-- 
Alexey Kurov <nucleo <at> fedoraproject.org>
alekcejk | 1 Jan 2012 23:15

Re: Sound disappears if "Enable echo cancellation" disabled

Sorry, I forgot to point that I am using linphone-3.5.0 in Fedora 16.
It is available now in F16 updates-testing repository and can
be updated with
"yum --enablerepo updates-testing update linphone ortp"

> Hi,
> 
> I have a problem with sound if "Enable echo cancellation" option disabled.
> In some calls sound disappears when this option disabled but volume
> indicators shows sound volume as if sound was fine.
> Sound appears after Pause/Resume pressed.
> There are no any unusual messages in debug window.
> No such problem if "Enable echo cancellation" option is enabled.
> 
> I also noticed that after I press button that disables microphone
> and then press Pause/Resume then button looks as if microphone
> was not disabled but actually it is disabled.
> 
--

-- 
Alexey Kurov <nucleo <at> fedoraproject.org>
Goran Kopcak | 2 Jan 2012 14:05
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Favicon

Linphone on tablets

Hi,

 

Is it possible to run video on any market available Android tablets (Samsung Galaxy Tab 10.1 preferably)?

 

BR,

Goran

 

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Koos Pol | 2 Jan 2012 18:16
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Changing my email address

Hi,

I'm trying to change my email address for my account on the free
Linphone SIP service.
After punching the buttons, I receive an activation link. But when I
click it, I get the attached error message.

Has anyone else tried to change their email address? Did it work?

Thanks,
Koos

Met vriendelijke groet,
S.C. Pol

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Liviu Andronic | 3 Jan 2012 00:08

SRTP encrypted calls error

Dear all
Is it normal that when I'm selecting 'SRTP' encryption and making a call  
to, say, <sip:echo <at> iptel.org>, I get the following error:
"No codec intersection"

I find the following message in teh log:
"error: Incompatible SDP offer received in 200Ok, need to abort the call".

Using Linphone 3.5.0 on Ubuntu Lucid, compiled against 'srtp-1.4.4'.

Thank you
Liviu

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Guillaume BERAUDO | 3 Jan 2012 08:33

Re: SRTP encrypted calls error

Hi Liviu,

Thanks for your report.

Could you send more logs?
In particular I would like to see the INVITE sent and the associated
200Ok response.

Cheers,

Guillaume Beraudo

Le Tue, 03 Jan 2012 00:08:15 +0100,
"Liviu Andronic" <liv-public <at> myopera.com> a écrit :

> Dear all
> Is it normal that when I'm selecting 'SRTP' encryption and making a
> call to, say, <sip:echo <at> iptel.org>, I get the following error:
> "No codec intersection"
> 
> I find the following message in teh log:
> "error: Incompatible SDP offer received in 200Ok, need to abort the
> call".
> 
> Using Linphone 3.5.0 on Ubuntu Lucid, compiled against 'srtp-1.4.4'.
> 
> Thank you
> Liviu
> 
> 
Liviu Andronic | 3 Jan 2012 08:43

Re: SRTP encrypted calls error

On Tue, 03 Jan 2012 08:33:49 +0100, Guillaume BERAUDO  
<guillaume.beraudo <at> belledonne-communications.com> wrote:

> Hi Liviu,
>
> Thanks for your report.
>
> Could you send more logs?
> In particular I would like to see the INVITE sent and the associated
> 200Ok response.
>
Sent off-list. Regards
Liviu

> Cheers,
>
> Guillaume Beraudo
>
>
> Le Tue, 03 Jan 2012 00:08:15 +0100,
> "Liviu Andronic" <liv-public <at> myopera.com> a écrit :
>
>> Dear all
>> Is it normal that when I'm selecting 'SRTP' encryption and making a
>> call to, say, <sip:echo <at> iptel.org>, I get the following error:
>> "No codec intersection"
>>
>> I find the following message in teh log:
>> "error: Incompatible SDP offer received in 200Ok, need to abort the
>> call".
>>
>> Using Linphone 3.5.0 on Ubuntu Lucid, compiled against 'srtp-1.4.4'.
>>
>> Thank you
>> Liviu
>>
>>
>

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Guillaume BERAUDO | 3 Jan 2012 11:02

Re: SRTP encrypted calls error

Hi Liviu,

Linphone invites the remote part with SRTP information:
m=audio 7078 RTP/SAVP 110 111 112 3 8 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYY

But the answer sent by the iptel Sip Express Media Server doesn't contain
SRTP information:
m=audio 10576 RTP/AVP 110 3 8 0 101

That is why the call fails.

The behaviour expected by Linphone is that the remote part refuse call if it
cannot handle SRTP. Then Linphone would try a non encrypted invite.

Right now, your only solution is to disable SRTP.
On the Linphone side, I will update the "no codec intersection" UI error message
to a more general one.

Cheers,

Guillaume

RFC 4568 - 5.1.2: 
   When the answerer receives the initial offer with one or more crypto
   attributes for a given unicast media stream, the answerer MUST either
   accept exactly one of the offered crypto attributes, or the offered
   stream MUST be rejected.

> >> Dear all
> >> Is it normal that when I'm selecting 'SRTP' encryption and making a
> >> call to, say, <sip:echo <at> iptel.org>, I get the following error:
> >> "No codec intersection"
> >>
> >> I find the following message in teh log:
> >> "error: Incompatible SDP offer received in 200Ok, need to abort the
> >> call".
> >>
> >> Using Linphone 3.5.0 on Ubuntu Lucid, compiled against
> >> 'srtp-1.4.4'.
MJ Ray | 3 Jan 2012 16:06
Gravatar

refer to another sip address from the GUI?

I see that the command-line client linphonec has a "refer" command.
Is this feature available in the GTK GUI anywhere?

Also, is it possible to autoanswer with a refer?  I'm thinking of
temporarily redirecting to another phone, without having to play
with the provider's web admin.

Thanks for any tips,
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