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Diffstat (limited to 'sipsim/uac_media.xml')
| -rw-r--r-- | sipsim/uac_media.xml | 92 |
1 files changed, 92 insertions, 0 deletions
diff --git a/sipsim/uac_media.xml b/sipsim/uac_media.xml new file mode 100644 index 0000000..51b5433 --- /dev/null +++ b/sipsim/uac_media.xml @@ -0,0 +1,92 @@ +<?xml version="1.0" encoding="ISO-8859-1" ?> +<!DOCTYPE scenario SYSTEM "sipp.dtd"> + +<!-- This program is free software; you can redistribute it and/or --> +<!-- modify it under the terms of the GNU General Public License as --> +<!-- published by the Free Software Foundation; either version 2 of the --> +<!-- License, or (at your option) any later version. --> +<!-- --> +<!-- This program is distributed in the hope that it will be useful, --> +<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> +<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> +<!-- GNU General Public License for more details. --> +<!-- --> +<!-- You should have received a copy of the GNU General Public License --> +<!-- along with this program; if not, write to the --> +<!-- Free Software Foundation, Inc., --> +<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> +<!-- --> +<!-- Based on sipp default 'uac' scenario. --> +<!-- To make one call to test@10.0.2.15 from IP 10.0.2.20 using PCMU codec: + ./sipp -sf uac_media.xml 10.0.2.15 -i 10.0.2.20 -bind_local -m 1 -s test -key codec PCMU/8000 -key pt 99 + --> +<!-- --> + +<scenario name="Codec test"> + <send retrans="500"> + <![CDATA[ + INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]> + Call-ID: [call_id] + CSeq: 1 INVITE + Contact: sip:sipp@[local_ip]:[local_port] + Max-Forwards: 70 + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=- 42 42 IN IP[local_ip_type] [local_ip] + s=- + c=IN IP[media_ip_type] [media_ip] + t=0 0 + m=audio [media_port] RTP/AVP [pt] + a=rtpmap:[pt] [codec] + a=recvonly + ]]> + </send> + + <recv response="100" optional="true"> + </recv> + + <recv response="180" optional="true"> + </recv> + + <recv response="183" optional="true"> + </recv> + + <recv response="200" rtd="true"> + </recv> + + <send> + <![CDATA[ + ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] + Call-ID: [call_id] + CSeq: 1 ACK + Contact: sip:sipp@[local_ip]:[local_port] + Max-Forwards: 70 + Content-Length: 0 + ]]> + </send> + + <!-- wait for callee to hang up --> + <recv request="BYE"> + </recv> + + <send> + <![CDATA[ + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:[local_ip]:[local_port];transport=[transport]> + Content-Length: 0 + ]]> + </send> +</scenario> |