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

[kaleem rehman]

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

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