Unexpected call termination when using pjsip and srtp

Asterisk Asterisk General
1

Hi, I’m seeing calls consistently terminating ~29-30mins into the call (say to the Asterisk echo()) when using chan_pjsip (with TLS) and SRTP (media_encryption=sdes). If I change pjsip.conf to use RTP (media_encryption=no), or use chan_sip, all works as expected.

Environment: Centos-6 64-bit, Asterisk-13.7.2, PJSIP-2.4.5, libsrtp (1.5.2).

Anyone else seeing similar problems, or can confirm it works beyond 30mins in their PJSIP + SRTP setup?

Thanks in advance.

2

As I’ve requested on the issue[1] you created the logs are needed before any comments can really be made.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-26003