Brekeke PBX Wiki

SRTP Support

 

Brekeke PBX version 3.2 and later:

Brekeke PBX supports SRTP. The following steps show how to enable SRTP for the phone assigned to a Brekeke PBX user extension.

1. From Brekeke PBX Admintool > [Options] > [Phone Type] page, create a new phone type

2. At new phone type setting page,

– set [RTP Relay] “on”

– select proper setting at [SRTP] field

– set proper Audio codecs in [Codec Priority] field.

3. Save the new phone type and restart Brekeke PBX from admintool.

4. From a PBX user extension > [Phones] page, select the phone type created above at the [Type] field under the phone which needs to enable SRTP for the call. And save the changes.

5. Making calls from/to this user with SRTP feature enabled in phone type, Brekeke PBX will do the conversion between the users who donot support SRTP.

SRTP support can also enabled:

– ARS route valid for the calls through the route

– Brekeke PBX Admintool > [Options] > [Settings] page, valid for all the calls through the PBX system.

 

* The following cipher suites are supported for SRTP on Brekeke PBX.

AES_CM_128_HMAC_SHA1_80   (Default)
AES_CM_128_HMAC_SHA1_32

 


Brekeke PBX version 3.1.x and earlier:

Brekeke PBXdoes not support SRTP. However, you can pass SRTP strem through SIP Server by using the following Dial Plan to bypass SRTP.

[Matching Patterns:]
$request = ^INVITE
To = sip:9(.+)@ 

[Deploy Patterns:]
To = sip:%1@

If a caller dials a number with the prefix “9”, the call will be forwarded to UA which supports SRTP directly. For example, if the UA’s number is “12345”, please dial “912345”

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