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
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@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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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 <jmmbuthia@gmail.com> *To:* freeswitch-users@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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depend the transcoding, from 10 to 20 ----- Original Message ----- From: James Mbuthia To: FreeSWITCH Users Help Sent: Monday, December 06, 2010 10:23 AM Subject: Re: [Freeswitch-users] SIP Phone Integration 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@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@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 -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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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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@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@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@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 ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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 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@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@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@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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already been done, check out red5phone. i have it working w/ FS and red5 server On Mon, Dec 6, 2010 at 9:49 AM, James Mbuthia <jmmbuthia@gmail.com> wrote:
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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participants (4)
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David Ponzone
-
James Mbuthia
-
Madovsky
-
vip killa