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[bullehs at gmail.com]

Dear all,

I have been testing an Opensips + Freeswitch setup on the same instance on EC2. I did follow the guide here for IP setup: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2

However, for production, it seems better to have separate instances for Opensips and Freeswitch. (Or wrong?). What would be the following variables:

bind_server_ip
external_rtp_ip
external_sip_ip

Shall I use the Opensips external IP or Freeswitch (although Freeswitch will have no external access).

Thanks for the help.

[david.villasmil.work a...]

If freeswitch has no external (public) access, and you don’t have an rtpproxy/rtpengine, and you will be servicing clients on the public internet, the audio will no work.


Your best bet for a simple setup is assign a public ip to freeswitch, allow all rtp port range (UDP) in and out of freeswitch, and set this public ip in external_rtp_ip. Everything else are the private IPs. 

On Sat, 5 Jun 2021 at 07:30, HS <bullehs@gmail.com (bullehs@gmail.com)> wrote:

[bullehs at gmail.com]

David.

Thanks a lot for taking the time to explain. I think in my hurry to post I forgot to add details.

Instance 1
Opensips + RTPProxy (Registrar + routing).
Instance 2
Freeswitch (IVR/Conferencing/VoiceMail etc.)

Does that setup make sense? What would be the variables (bind_server_ip, external_rtp_ip, external_sip_ip) be please?


Thanks again.

[david.villasmil.work a...]

You need to configure rtpproxy in multihomed mode (bridge mode), so it does external/internal relay of rtp. Then opensips will instruct rtpproxy to do it. 


On freeswitch just configure the private IPs.


That’s really all there is to it. If it’s configured correctly, you should see in the SDPs the following:


- Client—[INVITE]—>opensips: the client will send its public IP (or maybe its private, but if opensips is doing NAT translation, shouldnt matter, but I suggest testing with a client which sends its public IP)


- Opensips—>[INVITE]—>freeswitch: opensips should send rtpproxy’s private IP


The on the 200 OKs:


- Freeswitch—[200 OK]—>opesnsips: fs should send its private IP


- Opensips—[200 OK]—>client: opensips should send rtpproxy’s public IP address.


If you don’t see exactly this, something mis-configured somewhere.


Hope this help.

On Sat, 5 Jun 2021 at 15:41, HS <bullehs@gmail.com (bullehs@gmail.com)> wrote:

[freeswitch-users at li...]

------ Start of attached email. Subject: Re: [Freeswitch-users] Amazon EC2 Instances - Freeswitch ------
A simple and secure workaround, use a VPN like OpenVPN. No open ports,
no proxies, and it's secure and fast. Works on desktop, mobiles, etc.

Install OpenVPN in your server, and each client. Today every sip phone
can use openvpn in its setup.

Regards

El 05/06/2021 a las 03:29, HS escribió:

[freeswitch-users at li...]

------ Start of attached email. Subject: Re: [Freeswitch-users] Amazon EC2 Instances - Freeswitch ------
A simple and secure workaround, use a VPN like OpenVPN. No open ports,
no proxies, and it's secure and fast. Works on desktop, mobiles, etc.

Install OpenVPN in your server, and each client. Today every sip phone
can use openvpn in its setup.

Regards

PD: Sorry duplicate email, because previous wrong sended



El 05/06/2021 a las 03:29, HS escribió:

[bullehs at gmail.com]

Hi again.


David - thx a lot, will try and revert with issues. I just saw that you are on the Opensips list also

Norman - thx for the suggestion. However, I am quite new to this - unsure what problem OpenVPN is meant to solve in my scenario.


Best wishes.

[thesipguy at gmail.com]

I have a similar setup, I have OpenSIPs server without a rtp engine, my Asterisk/Freeswitch servers are on public IP's but port 5060 is blocked and can only receive traffic from the OpenSIPs IP's but the RTP ports are open on the Asterisk and Freeswitch servers, this way I avoid another hop on the audio, another benefit is that I can have Asterisk/Freeswitch servers in different parts of the world and route local traffic through them without needing to travel half the world to my OpenSIPs server and back.

On Sat, Jun 5, 2021, 09:47 HS <bullehs@gmail.com (bullehs@gmail.com)> wrote:

[itadebayo at gmail.com]

Hi Rosenber,

We have been trying to get OpeSIPS to work with freeswitch without success for some time now.


Can you share a basic configuration we use ASTPP.


Thanks.


Ismail
Rosenber,






[/url] Virus-free. [url=https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail]www.avast.com [url=#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2][/url]

On Tue, Jun 8, 2021 at 7:12 PM S.Rosenberg <thesipguy@gmail.com (thesipguy@gmail.com)> wrote:

[bullehs at gmail.com]

Pray tell Mr. Rosenberg

[thesipguy at gmail.com]

It depends a lot on your design, you need to know what you want OpenSIPs to do and what do you want Freeswitch to do, here is one tutorial by my friend  Giovanni Maruzzelli, he wrote the Freeswitch book and is very active on this forum.

https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration


On Wed, Jun 9, 2021 at 10:44 AM I. Adebayo <itadebayo@gmail.com (itadebayo@gmail.com)> wrote:

[itadebayo at gmail.com]

Thanks.

We go through the tutorial again.


Thanks.


Ismail




On Sun, Jun 13, 2021 at 4:41 PM S.Rosenberg <thesipguy@gmail.com (thesipguy@gmail.com)> wrote:

[bullehs at gmail.com]

My turn to help this time.


So I followed the following tutorial on one instance of Amazon EC2 and Opensips 3.0 (+RTPProxy):


https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration



There were some changes in the .cfg file - which I did over come (or so I think). The second was an error that showed up in the xml_handler.lua file.


Calls get to an IVR, however, I did have the following issues (and remain unsolved):


1. IVR/calls to Freeswitch disconnect after 30-32s.
2. Unsure if users are being pulled from the database or the profiles.
3. DTMF input isn't working (system doesn't respond to input.)



Hope that helps and maybe someone has a solution.

[bullehs at gmail.com]

Hello all,

David - worked with your suggestions for setting up the production environment (separate Opensips + Freeswitch instances). Most seems to work well. Except, in the tcpdump, I see when I call the IVR, the reply is being sent to the correct IP but the port number is 7078.


Any thoughts on the problem? And fix? Or more details required?



Thanks.

[david.villasmil.work a...]

The client is probably sending from 7078 and that’s why the reply is being sent there?

On Sun, 27 Jun 2021 at 17:24, HS <bullehs@gmail.com (bullehs@gmail.com)> wrote: