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

Fri Dec 3 19:16:34 UTC 2021

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 at lists.freeswitch.org> on behalf of kaleem rehman <k4kaleem at gmail.com>
Sent: Friday, December 3, 2021 5:08 AM
To: freeswitch-users at 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 at gmail.com<mailto: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
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From: kaiduan xie <kaiduanx at yahoo.ca<mailto:kaiduanx at yahoo.ca>>
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
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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<mailto: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<mailto:ehtasham.malik at expertflow.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Ahmed Hasan <ahmad.hasan at expertflow.com<mailto: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. )

[https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLClQgesUimaC5FBhoqcjL_0TiGnF8-z4kSqJywWlkYPcySl1rLcS18hzt1PbCqtGbvbtl6TVHQlFadPziihRZ17vCkCMArghS] 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 at expertflow.com<mailto:andreas.stuber at expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/> shani.awan3

On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem at gmail.com<mailto: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<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 at freeswitch.com<mailto:sales at 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
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