TLS SRTP Supports and Session Management by external API

kingston · Brekeke Member Joined: 16 Oct 2006 Posts: 13

1. Brekeke Product Name and version:

Brekeke SIP Server 2.0.7.2/217

2. Java version:

1.5.0_11

3. OS type and the version:

Linux 2.6.9-42.ELsmp

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:

Hi,

Does Brekeke SIP Server v2.0x support Secure RTP (SRTP) and SIP Transport Layer Security (TLS) Transportation? Any plan to put such features in the upcoming version?

Is there any APIs or workaround available for session management purpose? Such that I can able to retrieve the session status or manually terminate the the active calls without the WEB GUI.

Thanks.

Regards,
Kingston

voipwell.com · Posted: Wed Jun 06, 2007 10:56 pm Post subject:

Hi,

Don't know the answer to question 1 but I think you are looking for the new pal feature of the pbx to be able to control sessions from a windows machine with a new development kit. Here is the download page and be sure to check out the pdf link for the manual. This looks really hot!

http://www.brekeke.com/products/products_pbx_pal.php

kingston · Brekeke Member Joined: 16 Oct 2006 Posts: 13

Hi,

Thank you for your advice. However, PBX Active Library (PAL) can only support Microsoft Windows platform. Is there any development kit or Java API that is OS independent but do the same functionality, i.e. develop an application for session management in Brekeke SIP server?

Besides, I am keen to look for the solution to make Brekeke Server supports TLS and SRTP transportaion. Any suggestion how I implement these secure communication features in SIP server? Thanks.

Regards,
Kingston

Peely · Partner PBX Joined: 07 Sep 2005 Posts: 26

BSS does not support Transport Layer Security and therefore does not support secure RTP or SIPS.

Regards,

Neil.

Peely · Partner PBX Joined: 07 Sep 2005 Posts: 26

WRT your query regarding session management and call disconnection. This could be achieved using the accounting plugin and dial-plan plugin features provided in V2 upwards.

The accounting plugin can tell you when users initiate, start talking (200) and when the session tears down. You could store this in a database to determine the various call states.

The dial-plan plugin would allow you to determine the talktime for any given session based on the origination and destination, or anything else contained within the SIP packet of the INVITE.

Regards,

Neil.