Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
Hi Kaiduan, thanks for looking into it. Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent. no need for user to login to enter credentials, we want simple "*call us"* type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS. Regards, K ---------- Forwarded message ---------- From: kaiduan xie <kaiduanx@yahoo.ca> To: "freeswitch-users@lists.freeswitch.org" <freeswitch -users@lists.freeswitch.org> Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < k4kaleem@gmail.com> wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Ahmed Hasan <ahmad.hasan@expertflow.com> Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik@expertflow.com <andreas.stuber@expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com> wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Hi All, any takes on this plz. On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman <k4kaleem@gmail.com> wrote:
Hi Kaiduan,
thanks for looking into it.
Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent.
no need for user to login to enter credentials, we want simple "*call us"* type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS.
Regards, K ---------- Forwarded message ---------- From: kaiduan xie <kaiduanx@yahoo.ca> To: "freeswitch-users@lists.freeswitch.org" <freeswitch -users@lists.freeswitch.org> Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS?
/Kaiduan
On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < k4kaleem@gmail.com> wrote:
Salaam Ehtasham,
we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to.
Regards, Kaleem
---------- Forwarded message ---------- From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Ahmed Hasan <ahmad.hasan@expertflow.com> Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ?
Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. )
WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik@expertflow.com <andreas.stuber@expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3*
On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com> wrote:
Hi All,
our requirement is simple, we will have CALL US button on website
when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server.
With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end.
any ideas of achieving this Thanks, Kaleem _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
So the incoming request, Verto, WebRTC, SIPJS, whatever still gets authenticated with whatever credentials the web page supplies. So you could set up anonymous registration, and validate the credentials in the dial plan. You could dynamically validate the user and password and use those as tokens. You could also enforce only certain CODECs, for instance Opus, and anyone not using any of those would weed out most scripts. These measures, and Fail2Ban will prevent some unauthorized access but won’t help with DDoS or anyone actively looking to cause trouble (if an authentication token is provided by HTTPS its trivial to grab that if someone really wants to be malicious). Most other options would be expensive (hide behind CloudFlare) or onerous (use a CAPTCHA as authentication). It comes down to balancing requirements. If a client asked this from me we’d propose a one-time code provided on a Verto client that had a ten second timeout for login. The issue is ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of kaleem rehman <k4kaleem@gmail.com> Sent: Friday, December 3, 2021 5:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi All, any takes on this plz. On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman <k4kaleem@gmail.com<mailto:k4kaleem@gmail.com>> wrote: Hi Kaiduan, thanks for looking into it. Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent. no need for user to login to enter credentials, we want simple "call us" type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS. Regards, K ---------- Forwarded message ---------- From: kaiduan xie <kaiduanx@yahoo.ca<mailto:kaiduanx@yahoo.ca>> To: "freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman <k4kaleem@gmail.com<mailto:k4kaleem@gmail.com>> wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com<mailto:ehtasham.malik@expertflow.com>> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Ahmed Hasan <ahmad.hasan@expertflow.com<mailto:ahmad.hasan@expertflow.com>> Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) [https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLCl...] WWW:[domain2.png].expertflow.com<http://www.expertflow.com/> FB: [FB-f-Logo__blue_29.png] /Expertflow<https://www.facebook.com/Expertflow> LinkedIn: [linkedIn.png] /company/expertflow<https://www.linkedin.com/company/expertflow> Youtube: [YouTube-social-square_red_128px.png] /user/expertflow<https://www.youtube.com/user/expertflow> Twitter: [twitter.JPG] /Expertflow<https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik@expertflow.com<mailto:andreas.stuber@expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/> shani.awan3 On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com<mailto:k4kaleem@gmail.com>> wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community<https://signalwire.community/> Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com<https://freeswitch.com/> Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com<https://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 https://freeswitch.com<https://freeswitch.com/> _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community<https://signalwire.community/> Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com<https://freeswitch.com/> Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com<https://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 https://freeswitch.com<https://freeswitch.com/>
hi Thanks for ur msg, it was chopped towards the end says: ""the issue is" and nothing after. this gives me idea on solution. on verto demo, i can see it asks for login and then extension to make call. if i want all this hidden from user and in background. so end users sees a button only in app and it says "call us" once clicked everything is in background like logging softphone and starting call. all users hear is freeswitch answering and playing relative IVR script. what would be best way to achieve this. also, from Verto side, if i want to attach some data with call. what method i need to use to achieve. thanks, k On Fri, Dec 3, 2021 at 7:17 PM Support from NetworkedAudio LLC < support@naud.io> wrote:
So the incoming request, Verto, WebRTC, SIPJS, whatever still gets authenticated with whatever credentials the web page supplies.
So you could set up anonymous registration, and validate the credentials in the dial plan.
You could dynamically validate the user and password and use those as tokens. You could also enforce only certain CODECs, for instance Opus, and anyone not using any of those would weed out most scripts.
These measures, and Fail2Ban will prevent some unauthorized access but won’t help with DDoS or anyone actively looking to cause trouble (if an authentication token is provided by HTTPS its trivial to grab that if someone really wants to be malicious).
Most other options would be expensive (hide behind CloudFlare) or onerous (use a CAPTCHA as authentication). It comes down to balancing requirements.
If a client asked this from me we’d propose a one-time code provided on a Verto client that had a ten second timeout for login.
The issue is ------------------------------ *From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of kaleem rehman <k4kaleem@gmail.com> *Sent:* Friday, December 3, 2021 5:08 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
Hi All,
any takes on this plz.
On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman <k4kaleem@gmail.com> wrote:
Hi Kaiduan,
thanks for looking into it.
Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent.
no need for user to login to enter credentials, we want simple "*call us"* type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS.
Regards, K ---------- Forwarded message ---------- From: kaiduan xie <kaiduanx@yahoo.ca> To: "freeswitch-users@lists.freeswitch.org" <freeswitch -users@lists.freeswitch.org> Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS?
/Kaiduan
On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < k4kaleem@gmail.com> wrote:
Salaam Ehtasham,
we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to.
Regards, Kaleem
---------- Forwarded message ---------- From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Ahmed Hasan <ahmad.hasan@expertflow.com> Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ?
Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. )
WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik@expertflow.com <andreas.stuber@expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3*
On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com> wrote:
Hi All,
our requirement is simple, we will have CALL US button on website
when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server.
With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end.
any ideas of achieving this Thanks, Kaleem _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
participants (2)
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kaleem rehman
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Support from NetworkedAudio LLC