[Freeswitch-users] SIP Phone Integration

[David Ponzone]

Acme company ?

You meant ACME Packet ?

A ACME SBC just for transcoding would be a little bit overkill, except if he really wants to be able to go up to 30,000 concurrent calls.
And I am pretty sure FreeSWITCH with 15 D500 cards maxed out could achieve the same result, for a fraction of the price.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

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Le 06/12/2010 à 19:01, Madovsky a écrit :

> or use acme company,
> but very expensive
> ----- Original Message -----
> From: David Ponzone
> To: FreeSWITCH Users Help
> Sent: Monday, December 06, 2010 12:56 PM
> Subject: Re: [Freeswitch-users] SIP Phone Integration
> 
> If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
> According their website, they don't support Speex.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
> 
> 
> 
> 
> Le 06/12/2010 à 16:23, James Mbuthia a écrit :
> 
>> Thanks,  about how many concurrent calls are you able to when transcoding with FS?
>> 
>> On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:
>> You can transcode with FS,
>> I did already a web phone (boophone) with FS but ;keep in mind
>> that transcode=CPU
>> ----- Original Message -----
>> From: James Mbuthia
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Monday, December 06, 2010 9:49 AM
>> Subject: [Freeswitch-users] SIP Phone Integration
>> 
>> Hi guys,
>> 
>> Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone
>> 
>>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 
>> 
>> My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.
>> 
>> regards,
>> James Mbuthia
>> 
>> 
>> 
>> 
>> 
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