Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-call / 150_srtp_3_1.py
1 # $Id$
2 #
3 from inc_cfg import *
4
5 test_param = TestParam(
6                 "Callee=optional (with duplicated offer) SRTP, caller=optional SRTP",
7                 [
8                         InstanceParam("callee", "--null-audio --use-srtp=3 --srtp-secure=0 --max-calls=1"),
9                         InstanceParam("caller", "--null-audio --use-srtp=1 --srtp-secure=0 --max-calls=1")
10                 ]
11                 )