Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-call / 300_ice_1_1.py
1 # $Id$
2 #
3 from inc_cfg import *
4
5 # ICE mismatch
6 test_param = TestParam(
7                 "Callee=use ICE, caller=use ICE",
8                 [
9                         InstanceParam("callee", "--null-audio --use-ice --max-calls=1", enable_buffer=True),
10                         InstanceParam("caller", "--null-audio --use-ice --max-calls=1", enable_buffer=True)
11                 ]
12                 )