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

Fri Dec 3 22:52:54 UTC 2021

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 at 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 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> 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. )
>>
>> 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 at expertflow.com
>> <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> 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 at 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 at lists.freeswitch.org
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>> 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 at 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 at 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 at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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