Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 001_torture_4475_3_1_1_4.py
1 # $Id$
2 import inc_sip as sip
3 import inc_sdp as sdp
4
5 # Torture message from RFC 4475
6 # 3.1.1.  Valid Messages
7 # 3.1.1.4. Escaped Nulls in URIs
8 complete_msg = \
9 """REGISTER sip:example.com SIP/2.0
10 To: sip:null-%00-null@example.com
11 From: sip:null-%00-null@example.com;tag=839923423
12 Max-Forwards: 70
13 Call-ID: escnull.39203ndfvkjdasfkq3w4otrq0adsfdfnavd
14 CSeq: 14398234 REGISTER
15 Via: SIP/2.0/UDP host5.example.com;rport;branch=z9hG4bKkdjuw
16 Contact: <sip:%00@host5.example.com>
17 Contact: <sip:%00%00@host5.example.com>
18 L:0
19 """
20
21
22 sendto_cfg = sip.SendtoCfg( "RFC 4475 3.1.1.4", 
23                             "--null-audio --auto-answer 200", 
24                             "", 200, complete_msg=complete_msg)
25