Verto - audio only between FS and client
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! Hoping for some ideas or guidance. Thanks R
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
That would be open to someone changing the code and gaining access though
On 12 Mar 2018, at 18:59, Michael Jerris <mike@jerris.com> wrote:
in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
There are granular security functions available on the server side. Exploring exactly what you want available and not available is a bit outside of the scope of the support we can provide here, but we could help you out with a solution as part of a professional services offering.
On Mar 12, 2018, at 3:27 PM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
That would be open to someone changing the code and gaining access though
On 12 Mar 2018, at 18:59, Michael Jerris <mike@jerris.com> wrote:
in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Michael Jerris <mike@jerris.com> Sent: Monday, March 12, 2018 6:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/> FreeSWITCH<http://www.freeswitch.org/> www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg]<http://www.cluecon.com/> ClueCon Telephony and WebRTC Developer's conference<http://www.cluecon.com/> www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/> FreeSWITCH<http://www.freeswitch.org/> www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.
Yes, that’s excellent documentation, thanks for reminding me of that. How exactly did you use the tokens to ensure that people couldn’t invade other sessions, or was that not an issue in your context? ATM I have users connecting to conferences in the format of <conf_myname_token> as the conference name, but it’s proving quite tricky to keep track of, especially if someone logs on from two browsers. Ideally I’d like to just have users connecting to ‘nothing’, with the app transferring the channel to <conf_myname> (and no ability for the client to write their own code to force a connection to <conf_someoneelse>).
On 12 Mar 2018, at 19:35, Dom Rumsey <domrumsey@hotmail.com> wrote:
Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html <https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html>
I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously.
From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> on behalf of Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> Sent: Monday, March 12, 2018 6:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client
in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/> FreeSWITCH <http://www.freeswitch.org/> www.freeswitch.org <http://www.freeswitch.org/> FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.
http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/> ClueCon Telephony and WebRTC Developer's conference <http://www.cluecon.com/> www.cluecon.com <http://www.cluecon.com/> ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry.
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/> FreeSWITCH <http://www.freeswitch.org/> www.freeswitch.org <http://www.freeswitch.org/> FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
We generate a token every time a user joins a session (which is developed using a server side scripting language) - it can only be validated once. The token and the conference name is forwarded to a VertoJS client, which validates them using a REST API. If either of them is not validated the session fails. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Rick Jarvis <rick@magicmail.mooo.com> Sent: Monday, March 12, 2018 8:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client Yes, that’s excellent documentation, thanks for reminding me of that. How exactly did you use the tokens to ensure that people couldn’t invade other sessions, or was that not an issue in your context? ATM I have users connecting to conferences in the format of <conf_myname_token> as the conference name, but it’s proving quite tricky to keep track of, especially if someone logs on from two browsers. Ideally I’d like to just have users connecting to ‘nothing’, with the app transferring the channel to <conf_myname> (and no ability for the client to write their own code to force a connection to <conf_someoneelse>). On 12 Mar 2018, at 19:35, Dom Rumsey <domrumsey@hotmail.com<mailto:domrumsey@hotmail.com>> wrote: Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> on behalf of Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> Sent: Monday, March 12, 2018 6:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client in the source code for the client.
On Mar 12, 2018, at 2:53 PM, Rick Jarvis <rick@magicmail.mooo.com<mailto:rick@magicmail.mooo.com>> wrote:
How would I do that, in the verto config?
On 12 Mar 2018, at 18:38, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote:
you can just remove the features you don’t want from the client
On Mar 12, 2018, at 9:17 AM, Rick Jarvis <rick@magicmail.mooo.com<mailto:rick@magicmail.mooo.com>> wrote:
What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why:
My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server.
So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS.
The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code.
I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now!
Hoping for some ideas or guidance.
Thanks R
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> [https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/> FreeSWITCH<http://www.freeswitch.org/> www.freeswitch.org<http://www.freeswitch.org/> [https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/> FreeSWITCH<http://www.freeswitch.org/> www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> [https://cluecon.com/theme/img/cluecon/hook1.jpg]<http://www.cluecon.com/> ClueCon Telephony and WebRTC Developer's conference<http://www.cluecon.com/> www.cluecon.com<http://www.cluecon.com/> [https://cluecon.com/theme/img/cluecon/hook1.jpg]<http://www.cluecon.com/> ClueCon Telephony and WebRTC Developer's conference<http://www.cluecon.com/> www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> lists.freeswitch.org<http://lists.freeswitch.org/> lists.freeswitch.org Mailing Lists<http://lists.freeswitch.org/> lists.freeswitch.org lists.freeswitch.org Mailing Lists: Welcome! Below is a listing of all the public mailing lists on lists.freeswitch.org. Click on a list name to get more ... To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> [https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/> FreeSWITCH<http://www.freeswitch.org/> www.freeswitch.org<http://www.freeswitch.org/> FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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<http://www.freeswitch.org/>
Hello, We are looking to implement Freeswitch on an embedded ARM platform. This platform has 3 interfaces: - PSTN - USB with a 3g dongle - IP This platform will manage only one simultaneous call. From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations? Many thanks in advance, Regards, Bruno
Would be enough. Regards, Igor On Mar 13, 2018, 5:14 PM +0200, internet@hernandez.eu, wrote:
Hello,
We are looking to implement Freeswitch on an embedded ARM platform.
This platform has 3 interfaces: - PSTN - USB with a 3g dongle - IP
This platform will manage only one simultaneous call.
From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations?
Many thanks in advance,
Regards,
Bruno
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Freeswitch recommendations are hard to nail down due to the hugely varied uses. The best policy is to properly test all your use cases on the platform and see how it performs. It could very much depend on the hardware/software requirements for the 3g and pstn interfaces.
On Mar 13, 2018, at 5:30 AM, internet@hernandez.eu wrote:
Hello,
We are looking to implement Freeswitch on an embedded ARM platform.
This platform has 3 interfaces: - PSTN - USB with a 3g dongle - IP
This platform will manage only one simultaneous call.
From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations?
Many thanks in advance,
Regards,
Bruno
hi, works flawless for me on many customer raspberry pi devices Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform From: internet@hernandez.eu To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Date: 3/13/2018, 1:30:40 PM
Hello,
We are looking to implement Freeswitch on an embedded ARM platform.
This platform has 3 interfaces: - PSTN - USB with a 3g dongle - IP
This platform will manage only one simultaneous call.
From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations?
Many thanks in advance,
Regards,
Bruno
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
i like this one: https://www.pine64.org/?page_id=1194 T. On 13 March 2018 at 21:10, John Griessen <john@industromatic.com> wrote:
On 03/13/2018 11:01 AM, Bipin Patel wrote:
hi,
works flawless for me on many customer raspberry pi devices
What are other favorite ARM platforms for FS? Perhaps with more memory?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
any one knows if there are any embedded platform that has FXO/FXS interfaces? On Wed, Mar 14, 2018 at 2:10 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
i like this one: https://www.pine64.org/?page_id=1194
T.
On 13 March 2018 at 21:10, John Griessen <john@industromatic.com> wrote:
On 03/13/2018 11:01 AM, Bipin Patel wrote:
hi,
works flawless for me on many customer raspberry pi devices
What are other favorite ARM platforms for FS? Perhaps with more memory?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
participants (9)
-
Abaci B
-
Bipin Patel
-
Dom Rumsey
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Igor Olhovskiy
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internet＠hernandez.eu
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John Griessen
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Michael Jerris
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Rick Jarvis
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Tihomir Culjaga