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Franklin | 3 Jul 2003 03:53

two questions about linphonec


Hello: 

I am reading the console version of linphone--linphonec and have two 
questions need your help. 

1)When the user types in "c", linphonec uses linphone_core_invite to send 
sip message to the called party. But I can't see how the calling party wait 
for the message from the called party and which function it uses to open the

soundcard and send/receive the rtp stream. 
2) The are two kind of sound card linphonec support:oss card and Alsa. Thay 
are encapsulated in the osscard.c amd alsacard.c respectively. How wil 
linphonec decide which kind of card is in use and then used the API 
accordingly? 

Thank you very much in advance. 

Bestwishes 
Franklin
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yogesh | 8 Jul 2003 07:10
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ip-pstn gateway

Hello,
    I am looking for a sip server with IP-PSTN connectivity. Can anyone 
point to a location where i can find this ? Of course, a free  public 
server wil be my first choice. :)

thanks
Yogesh
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stefanfritzsche | 8 Jul 2003 11:30
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Picon

uncommon sip-signalling

Hello,

First of All:
Thanks for writing linphone.

I have a question about some special kinds
of SIP-signalling, and whether linphone
supports it.

The normal way of initiating a session
processes as following:

---------- Invite (with SDP) ---------->
<-------------- Trying -----------------
<-------------- Ringing ----------------
<--------- 200 OK (with SDP) -----------
----------------- ACK ----------------->

Ok, that works fine with linphone and
partysip.
(except of that anoying latency caused by
my old-school-soundcard's dsp-blocksize
set to 8192)

But there are other options of SIP-Signalling
mentioned in the RFCs.
For instance an ACK to 200 OK may contain an
application/sdp message body. This is permitted
if the initial INVITE did not contain an sdp
message body:
(Continue reading)

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Aymeric Moizard | 8 Jul 2003 11:40

Re: uncommon sip-signalling


Linphone will be ported to eXosip soon which will add support for on/hold
off/hold. (the body in the ACK might be still unsupported for some time!)

It will also add support for REFER, SUBSCRIBE, NOTIFY and MESSAGE
methods.

Aymeric

On Tue, 8 Jul 2003, stefanfritzsche <at> gmx.de wrote:

> Hello,
>
> First of All:
> Thanks for writing linphone.
>
> I have a question about some special kinds
> of SIP-signalling, and whether linphone
> supports it.
>
> The normal way of initiating a session
> processes as following:
>
> ---------- Invite (with SDP) ---------->
> <-------------- Trying -----------------
> <-------------- Ringing ----------------
> <--------- 200 OK (with SDP) -----------
> ----------------- ACK ----------------->
>
>
(Continue reading)

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Jamey Hicks | 8 Jul 2003 14:22
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Re: ip-pstn gateway

On Tue, 2003-07-08 at 01:10, yogesh wrote:
> Hello,
>     I am looking for a sip server with IP-PSTN connectivity. Can anyone 
> point to a location where i can find this ? Of course, a free  public 
> server wil be my first choice. :)
> 

We have not used the PSTN connectivity in Asterisk yet, but we've been
pretty happy with its SIP and voicemail capabilities. 

http://www.asteriskpbx.com/

We are using a sip express router (http://www.iptel.org/ser/) as SIP
registrar/proxy, with a Cisco 2600 to a Nortel PBX to a PRI as our PSTN
gateway.  We are considering simplifying that with a new phone line and
a PRI PCI card with asterisk because it would probably work better. 
btw, we are currently debugging a problem where the Cisco 2600 drops
calls from linphone because of a "server error".

-Jamey
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Jamey Hicks | 8 Jul 2003 14:31
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Re: uncommon sip-signalling

On Tue, 2003-07-08 at 05:30, stefanfritzsche <at> gmx.de wrote:

> Another way is, to put the callee "onhold"
> by using an SDP-body without media-lines
> and the connection address set to "0.0.0.0".
> 

One of my coworkers has just modified linphone to support this method
for putting the SIP UA on hold.  In addition, he has added support for
performing call transfers via either REFER or BYE with an Also field
(used by Cisco 7960 SIP phones).  For outgoing requests, it currently
only supports BYE/Also.  We would be happy to send a patch if you're
interested.

-Jamey
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Kannaiyan Natesan | 8 Jul 2003 15:06

Re: ip-pstn gateway

Do they windows version of it?

----- Original Message ----- 
From: "Jamey Hicks" <jamey.hicks <at> hp.com>
To: "yogesh" <yogeshd <at> aftek.com>
Cc: <linphone-users <at> nongnu.org>
Sent: Tuesday, July 08, 2003 1:22 PM
Subject: Re: [Linphone-users]ip-pstn gateway

> On Tue, 2003-07-08 at 01:10, yogesh wrote:
> > Hello,
> >     I am looking for a sip server with IP-PSTN connectivity. Can anyone
> > point to a location where i can find this ? Of course, a free  public
> > server wil be my first choice. :)
> >
>
> We have not used the PSTN connectivity in Asterisk yet, but we've been
> pretty happy with its SIP and voicemail capabilities.
>
> http://www.asteriskpbx.com/
>
> We are using a sip express router (http://www.iptel.org/ser/) as SIP
> registrar/proxy, with a Cisco 2600 to a Nortel PBX to a PRI as our PSTN
> gateway.  We are considering simplifying that with a new phone line and
> a PRI PCI card with asterisk because it would probably work better.
> btw, we are currently debugging a problem where the Cisco 2600 drops
> calls from linphone because of a "server error".
>
> -Jamey
>
(Continue reading)

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Jamey Hicks | 8 Jul 2003 16:41
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Favicon

Re: ip-pstn gateway

On Tue, 2003-07-08 at 09:06, Kannaiyan Natesan wrote:
> Do they windows version of it?
> 

There does not appear to be windows support for asterisk. 

-Jamey
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Hamid Diwan | 8 Jul 2003 18:43
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remove me


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Hubert Jin | 11 Jul 2003 04:14

question about r->qoutputs[0] !

Hi, all,

I am now trying to port sipomatic to embeded linux environment.

The problem I meet is that I can not get any voice  from sipomatic.  The 
sipomatic is running on the uClinux board, can spawn the third thread 
when it receives INVITE request for linphonec calling from PC. The 
signalling connection is OK and I can hear the Ring.

Through debugging I found that at line 133 of msread.c,   r->qoutputs[0] 
is always 0, so it always skips the following IF statement.  I am sure 
it is assigned a value before.

My question is under what condition will sipomatic clear r->qoutputs[0]? 
Can anyone give me some suggestions?

Thank in advance.

Hubert
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