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SIP Registation Forwarding using UDP, TCP, and TLS

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SIP Registation Forwarding using UDP, TCP, and TLS Empty SIP Registation Forwarding using UDP, TCP, and TLS

Post by jmross on Wed Mar 11, 2020 2:01 pm

Hi, I've followed the steps in the tutorial named "FreeSBC:Remote_Workers:Configuration_A" on the wiki and I seem to have it working where registrations and signaling is correctly forwarded to an Asterisk server.

In Asterisk using pjsip, it's easy to have multiple transports and listen on port 5060 using UDP and TCP and 5061 using TLS.

I'm having trouble configuring FreeSBC to do this.

I created several SIP transports, one for each protocol. Do I need to create a corresponding NAP and route for each?

ie.

OPEN_NAP_UDP -> SIP_SERVER_NAP_UDP
OPEN_NAP_TCP -> SIP_SERVER_NAP_TCP
OPEN_NAP_TLS -> SIP_SERVER_NAP_TLS

I ask because when adding multiple transports to a NAP, FreeSBC says "this is not supported, and may cause undefined behaviors.".

Do I need a separate SIP Registrar for each as well?

Thanks

jmross

Number of Messages : 9
Point : 19
Registration Date : 2020-03-06

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SIP Registation Forwarding using UDP, TCP, and TLS Empty Re: SIP Registation Forwarding using UDP, TCP, and TLS

Post by Admin on Thu Mar 12, 2020 2:33 am

Our SBC is not supporting TLS yet. You cannot use multiple transport in same NAP. You need to create different NAPs with different transports.

Admin
Admin

Number of Messages : 368
Point : 905
Registration Date : 2017-11-27

http://freesbc.yetkinforum.com

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Post by jmross on Thu Mar 12, 2020 9:28 am

I'm using FreeSBC 3.1.10 and this forum says that it does support TLS though:

================================
What's new in release 3.1.10
================================
Enhancements:
- Support SIP TCP and SIP TLS
- FQDN (fully qualified domain name) Enhanced functionality
- Raise SIP Transport server limit to 96
- Support SIPREC relay
- Microsoft Azure cloud platform

Fixes:
#23213 - DHCP MTU is hardcoded at 1500
#23282 - Single port interface should not be readonly
#23471 - Vmxnet3 hardware IP checksum not calculated for RTP packets
#23600 - Add call-id in the routing script
#23602 - FQDN SRV record support

jmross

Number of Messages : 9
Point : 19
Registration Date : 2020-03-06

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Post by Admin on Thu Mar 12, 2020 9:31 am

You are right. My mistake. We don't support SRTP yet. If you want to use SRTP with TLS, it will not work.

Admin
Admin

Number of Messages : 368
Point : 905
Registration Date : 2017-11-27

http://freesbc.yetkinforum.com

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Post by jmross on Thu Mar 12, 2020 2:59 pm

I think I got it working using two NAPs on the wan side just for UDP and TCP for now.

But I'm getting multiple registrations using TCP using the softphone Linphone. It registers fine at first, but every minute or so it tries again. These contacts pile up on FreeSBC and Asterisk until it hits the max contacts allowed on Asterisk.

SIP Registation Forwarding using UDP, TCP, and TLS Screen10

Do you know why this might be happening?

jmross

Number of Messages : 9
Point : 19
Registration Date : 2020-03-06

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Post by Admin on Fri Mar 13, 2020 12:59 am

Hi

I think you need to follow up the issue with support@telcobridges.com. Please send an email with all details.

Admin
Admin

Number of Messages : 368
Point : 905
Registration Date : 2017-11-27

http://freesbc.yetkinforum.com

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