Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 174_timer_se_too_small.py
1 # $Id$
2 import inc_sip as sip
3 import inc_sdp as sdp
4
5 sdp = \
6 """
7 v=0
8 o=- 0 0 IN IP4 127.0.0.1
9 s=pjmedia
10 c=IN IP4 127.0.0.1
11 t=0 0
12 m=audio 4000 RTP/AVP 0 101
13 a=rtpmap:0 PCMU/8000
14 a=sendrecv
15 a=rtpmap:101 telephone-event/8000
16 a=fmtp:101 0-15
17 """
18
19 pjsua_args = "--null-audio --auto-answer 200 --timer-min-se 2000 --timer-se 2000"
20 extra_headers = "Supported: timer\nSession-Expires: 1800\n"
21 include = ["Min-SE:\s*2000"]
22 exclude = []
23 sendto_cfg = sip.SendtoCfg("Session Timer SE too small", pjsua_args, sdp, 422, 
24                            extra_headers=extra_headers,
25                            resp_inc=include, resp_exc=exclude) 
26