RTP Encryption

Hello Guys.

I have one problem and it would be really cool if somebody can give me a tip to come over it :slight_smile:

I have one existing asterisk instance (1.4) with a lot of snom phones online, now i deployed one freepbx distro that i´m testing (latest stable).

snom phones have rtp encryption enabled by default, so if i test it with freepbx 1.8 latest stable i get an error as there is no rtp encrytpion configure on it i think.

the error show this:
[2013-07-23 18:04:26] DEBUG[1678]: sip/sdp_crypto.c:283 sdp_crypto_process: Accepting crypto tag 1
[2013-07-23 18:04:26] DEBUG[1678]: sip/sdp_crypto.c:313 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rLZ5oVDsLNWUE0b7x4y3kgHqBgOMhM4jyspE+eur
[2013-07-23 18:04:26] WARNING[1678]: chan_sip.c:9618 process_sdp: We are requesting SRTP for audio, but they responded without it!

of course one easy way to get it working is to disable rtp encryption on all phones, but because i have many phones even remote phones without direct access, it would be a big struggle to get the paramers on all phones configures.

as it works with asterisk 1.4 (enabled rtp encryption) i think this parameter just get´s ignored on it becuase 1.4 is maybe not capable for this.

my question is:
is it possible maybe to configure asterisk 1.8 freepbx distro anyhow to ignore this tag or maybe having another change central on the server in order to get those phone working without having the need to change every single phone?

thanks very much for any input.

kind regards,

This is a pain, and one of the reasons to always provision via tftp, ftp or http so you don’t have remote phones you can’t modify.

This is a complicated topic and the Asterisk discussion is in Jira: