[Freeswitch-users] WebRTC calls one way with custom sip messages UUI

[kaleem rehman]

Hi All,

any takes on this plz.

On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman <k4kaleem at 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 at yahoo.ca>
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch
> -users at 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 at 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 at expertflow.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: Ahmed Hasan <ahmad.hasan at 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. )
>
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>
>
> On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem at 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
> _________________________________________________________________________
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> _________________________________________________________________________
>
> 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 at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
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