[asterisk-dev] RTP/SAVP & TLS

[Joshua Colp]

Ross Beer wrote:
> Hi Dev,
>
> In Asterisk 1.8 Snom phones accept calls when RTP/SAVP is set to
> 'mandatory' which means that the RTP/SAVP options appear in the SDP 'm'
> lines. However in Asterisk 13 chan_pjsip, no such lines exist when using
> 'SDES' encryption.

The "media_encryption=sdes" option turns on SRTP support and thus makes 
the media RTP/SAVP. You can also turn on optimistic SRTP support as well 
using "media_encryption_optimistic=yes" which will use RTP/AVP but 
include a crypto line. I just checked the testsuite tests for SDP 
offer/answer and they are passing, I also manually enabled it and 
confirmed it is RTP/SAVP. You may have a configuration error.

>
> Therefore Snom phones require this option to be set to 'off'. Should
> Asterisk 13 be offering RTP/SAVP in the same way as chan_sip did?
>
> With regards to TLS, devices reject calls if a 'transport=transport-tls'
> is specified. Is this also a bug as it appears that Asterisk doesn't
> re-use an active connection in this situation?

This is a bug in PJSIP which has an issue on our side[1]. If an explicit 
transport is specified PJSIP will not reuse a connection.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-22658

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org