Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 301_srtp0_recv_savp.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=tester
10 c=IN IP4 127.0.0.1
11 t=0 0
12 m=audio 4000 RTP/SAVP 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 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WnD7c1ksDGs+dIefCEo8omPg4uO8DYIinNGL5yxQ
18 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:t0r0/apkukU7JjjfR0mY8GEimBq4OiPEm9eKSFOx
19 """
20
21 args = "--null-audio --auto-answer 200 --max-calls 1 --use-srtp 0 --srtp-secure 0"
22 include = []
23 exclude = []
24
25 sendto_cfg = sip.SendtoCfg( "Callee has SRTP disabled but receive RTP/SAVP, should reject the call", 
26                             pjsua_args=args, sdp=sdp, resp_code=488, 
27                             resp_inc=include, resp_exc=exclude)
28