概述
freeswitch是一款简单好用的VOIP开源软交换平台。
在更新了sipp模拟update的配置方案之后,我希望对比一下fs对update和reinvite的处理流程。
本文档记录sipp的配置方案,该方案中包含了update和reinvite的信令。
环境
CentOS 7.9
freeswitch 1.10.7
sipp.3.6.2
方案描述
测试环境模块。
eyebean --> fs-reg --> fs --> sipp
sipp作为uas端,模拟update和reinvite消息。
fs作为测试服务器,响应sipp的update和reinvite消息。
后续应该会针对fs的响应流程做优化。
sipp脚本
sipp的脚本中,分别模拟update消息和reinvite消息。
183消息中使用105作为rfc2833的payload。
update消息中使用103作为rfc2833的payload。
200接通之后,sipp会发送2个DTMF码(payload=101),fs会丢弃这2个码。
reinvite消息中使用101作为rfc2833的payload。
reinvite完成之后,sipp会再发送2个DTMF码(payload=101),fs会转发这2个码。
脚本 uas-test-reinvite.xml 内容如下。
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true" >
<action>
<ereg regexp=".*"
search_in="hdr"
header="From:"
check_it="true"
assign_to="1" />
<ereg regexp=".*"
search_in="hdr"
header="To:"
check_it="true"
assign_to="2" />
<ereg regexp="sip:[^;>]+"
search_in="hdr"
header="Contact:"
check_it="true"
assign_to="3" />
<ereg regexp=".*"
search_in="hdr"
header="CSeq:"
check_it="true"
assign_to="4" />
</action>
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=sippTag01
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1000"/>
<send>
<![CDATA[
SIP/2.0 183 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=sippTag01
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 [media_port] RTP/AVP 8 18 105
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:105 telephone-event/8000
a=fmtp:105 0-15
a=ptime:20
]]>
</send>
<pause milliseconds="1000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500" crlf="true">
<![CDATA[
UPDATE [$3] SIP/2.0
[last_Via:]
To: [$1]
From: [$2];tag=sippTag01
[last_Call-ID:]
CSeq: 11 UPDATE
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 [media_port] RTP/AVP 8 18 103
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-15
a=ptime:20
]]>
</send>
<recv response="200" crlf="true"> </recv>
<pause milliseconds="1000"/>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
From: [$1]
To: [$2];tag=sippTag01
[last_Call-ID:]
CSeq: [$4]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: 0
]]>
</send>
<recv request="ACK" crlf="true"> </recv>
<pause milliseconds="2000"/>
<nop>
<action>
<exec play_pcap_audio="../pcap/dtmf_2833_9.pcap"/>
</action>
</nop>
<pause milliseconds="500"/>
<nop>
<action>
<exec play_pcap_audio="../pcap/dtmf_2833_5.pcap"/>
</action>
</nop>
<pause milliseconds="1000"/>
<send retrans="500" crlf="true">
<![CDATA[
INVITE [$3] SIP/2.0
[last_Via:]
To: [$1]
From: [$2];tag=sippTag01
[last_Call-ID:]
CSeq: 12 INVITE
Max-Forwards: 64
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 [media_port] RTP/AVP 8 18 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" crlf="true">
</recv>
<send>
<![CDATA[
ACK [$3] SIP/2.0
[last_Via:]
To: [$1]
From: [$2];tag=sippTag01
[last_Call-ID:]
CSeq: 12 ACK
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 64
Content-Length: 0
]]>
</send>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="../pcap/dtmf_2833_2.pcap"/>
</action>
</nop>
<pause milliseconds="500"/>
<nop>
<action>
<exec play_pcap_audio="../pcap/dtmf_2833_7.pcap"/>
</action>
</nop>
<pause milliseconds="500"/>
<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>
<timewait milliseconds="3000"/>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
文件目录结构如下。
sipp.3.6.2/conf/uas-test-reinvite.xml
sipp.3.6.2/pcap/dtmf_2833_9.pcap
启动sipp-uas。
cd sipp.3.6.2/conf
sudo sipp -i 10.55.55.138 -p 5555 -sf uas-test-reinvite.xml
10011发起呼叫测试。
使用eyebean发起呼叫,并在10秒后挂机,查看fs节点对DTMF的处理情况。
sip信令和媒体截图
整个呼叫流程的sip信令和媒体截图。
总结
sipp很灵活,可以帮助我们在测试中构建各种模拟场景。
先比对update消息和reinvite消息的处理流程,再针对具体问题进行优化。
空空如常
求真得真
标签:last,ip,uas,Content,rtpmap,sipp,reinvite From: https://www.cnblogs.com/qiuzhendezhen/p/18412595