Hi all,
when i use sipp send the rtp to my softphone ,I find the rtp packets is
not arrive at the softphone in order that the softphone can not listen the
voice.
anyone can help me ,I appreciate very thanks!
jordan
the following is uac_pcap.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "[field0].dtd">
<!-- 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 distrib[field1]ed in the hope that it will be
useful, -->
<!-- b[field1] WITHO[field1] ANY WARRANTY; witho[field1] 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 -->
<!-- -->
<!-- [field0] default 'branchc' scenario.
-->
<!-- -->
<scenario name="\pcap play">
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: service <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="407" auth="true">
</recv>
<!-- By adding rrs="true" (Record Ro[field1]e Sets), the ro[field1]e
sets -->
<!-- are saved and used for following messages sent. Useful to test
-->
<!-- against stateful SIP proxies/B2BUAs.
-->
<!-- Packet lost can be simulated in any send/recv message by
-->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent.
-->
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: service <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: service <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication]
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="183"
optional="true">
</recv>
<recv response="180"
optional="true">
</recv>
<recv response="200" ctrf="true">
</recv>
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: service <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP
stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause milliseconds="100000"/>
<!-- This delay can be customized by the -d command-line option
-->
<!-- or by adding a 'milliseconds = "value"' option here.
-->
<!-- The 'crlf' option inserts a blank line in the statistics report.
-->
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: service <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
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