TRY THIS How to implement tls and srtp

[miguel]

I have it working On 1.8.23 but I was wondering if it is possible to install it On asterisk 11 any pointers can come in handy.

Best regards,

Miguel.

 

[miguel]

Anyone? Hasn't anybofy done this because i tried to compile the srtp wich i had done with 1.8.x and this didn't worked

 

[wardmundy]

If you just want to experiment, it's already built into our WebRTC Virtual Machine.

 

[miguel]

But I want to use it On a production machine with their software like bria (the one they use)

Best regards,

MN

 

[wardmundy]

Read the article!

 

[miguel]

Ward I have read the article but I wanted for them to use their own software I know it says that asterisk 11 is lacking of some of this feautures but asterisk 1.8 is lacking as well we have to install and compile the srtp module I have tried to do it On asterisk 11 but without luck to make it work, They wouldnt like to make their comunications via a web browser they want to make it via their own phones On their mobile or At Home with thier hadsets or voip Phones so what i wanted to know is if there is a way to implement it as I want to.

Best regards,

MN

 

[wardmundy]

The article has links to the SRTP methodology. You don't have to implement WebRTC if you don't want to.

 

[miguel]

Something must be wrong with me but i have re read it looking for links to srtp and could not find anything (feel very Bad) thank you anyway

 

[Porch]

I am using it on 11 just fine. I just used the instructions for 1.8.

 

[miguel]

Ok I did the same On the early versions of asterisk 11 and coukd not get it to work maybe something has changed, so you are telling me it works and it works well wuth the same method as 1.8?

 

[Porch]

Yep. I am running it on 11.4 and it's working fine. Finding a client that I liked, that properly supported it is a pain, but it's now working with the nightly of CSipSimple.

 

[miguel]

Ok so you compiled srtp module and everything?

 

[Porch]

Nope. I made the SSL cert using some how-to I can't find now. Then put the right lines into the config. srtp is compiled by default. You just have to set encryption=yes in the right conf file and it works.

Note that if you don't set encryption=yes, but have encryption turned on in the client, it srtp will still work for client->pbx connections. By default, PIAF seems to be configured for both RTP and SRTP incoming connections. But unless you set encryption=yes, it won't use SRTP for pbx->client connections and you won't get any audio.

 

[miguel]

Where do you put encryption=yes ? In sip settings?

 

[Porch]

In sip_custom_post.conf I have this for each extension.

[200](+)
transport=tls
encryption=yes

 

[miguel]

How to install srtp on raspi?