聊聊 SIPp uas.xml 及相关配置实践

杜金房

发布于 2026-06-15 16:58:59

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文章被收录于专栏：FreeSWITCH中文社区FreeSWITCH中文社区

SIPp 是一款常用的 SIP 协议性能测试工具，uas.xml 是其内置的用户代理服务器（UAS）场景配置文件。本文将从基础使用、代理适配修改、主动发送 BYE 消息配置三个维度，详细讲解 uas.xml 的相关操作。

获取 uas.xml 并编写启动脚本

首先，通过 SIPp 内置命令可直接导出默认 uas.xml 配置文件，执行以下命令即可：

sipp -sd uas > uas.xml

为了便捷启动 UAS 服务，我们可以编写一个启动脚本uas.sh，将启动参数固化其中。脚本内容如下，需根据实际环境修改 LOCAL_IP（本地 IP 地址）和 LOCAL_PORT（本地端口，默认 SIP 端口为 5060）：

#!/bin/bash

# 配置本地IP和端口（需根据实际环境调整）
LOCAL_IP=192.168.1.100
LOCAL_PORT=5060

# 启动SIPp UAS服务，指定配置文件、本地IP和端口
sipp -sf uas.xml -i $LOCAL_IP -p $LOCAL_PORT

脚本核心参数说明：

-sf：指定 SIPp 运行的场景配置文件（此处为 uas.xml）；
-i：绑定本地 IP 地址，用于接收 SIP 消息；
-p：绑定本地端口，用于监听 SIP 消息。

脚本编写完成后，需赋予执行权限并启动，命令如下：

# 赋予脚本执行权限
chmod +x uas.sh
# 启动 UAS
./uas.sh

启动成功后，FreeSWITCH 可直接呼叫 SIPp 监听的 IP 和端口，进行基础的 SIP 呼叫压力测试。

进阶：修改 uas.xml 以便适配 Kamailio 代理呼叫

若需通过 Kamailio（SIP 代理程序）实现 FreeSWITCH 与 SIPp 的呼叫交互，需对默认 uas.xml 进行三处关键修改，确保 SIP 消息能正确通过代理传递。

修改编码格式（支持中文注释）

默认 uas.xml 第一行编码为`ISO-8859-1`，不支持中文注释。将其修改为 `UTF-8` 编码，方便后续添加中文说明：

原内容：

<?xml version="1.0" encoding="ISO-8859-1" ?>

修改后：

<?xml version="1.0" encoding="UTF-8" ?>

接收 INVITE 消息时启用路由集保存

SIP 代理场景下，需保存路由集（Route Sets）以确保后续响应消息能正确回传给代理。找到接收 INVITE 消息的配置段，添加 rrs="true" 参数（rrs 即 Record Route Sets）：

原内容：

  <recv request="INVITE" crlf="true">
  </recv>

修改后：

  <recv request="INVITE" crlf="true" rrs="true">
  </recv>

说明：启用 rrs="true" 后，SIPp 会保存接收消息中的路由信息，用于后续发送响应/请求时使用，适配有状态 SIP 代理场景。

响应消息中添加 Record-Route 头域

Record-Route 头域用于记录呼叫路径，确保后续消息（如 BYE）能沿原路径回传。需在 180 Ringing 和 200 OK 两个响应消息中添加 `[last_Record-Route:]` 占位符（自动复用接收消息中的 Record-Route 信息）。

修改 180 Ringing 响应：

原内容：

 <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>

修改后（添加 [last_Record-Route:]）：

<send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>

修改 200 OK 响应（INVITE 最终响应）：

原内容：

  <send retrans="500">
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [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 0
      a=rtpmap:0 PCMU/8000
      a=ptime:20
    ]]>
  </send>

修改后（添加 [last_Record-Route:]）：

 <send retrans="500">
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      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 0
      a=rtpmap:0 PCMU/8000
      a=ptime:20
    ]]>
  </send>

适配代理的 uas.xml 完整内容

整合上述修改后，完整的 uas.xml 如下：

<?xml version="1.0" encoding="UTF-8" ?>
<!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             -->
<!--                                                                    -->
<!--                 Sipp default 'uas' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic UAS responder">
  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <!-- Adding ignoresdp="true" here would ignore the SDP data: that     -->
  <!-- can be useful if you want to reject reINVITEs and keep the       -->
  <!-- media stream flowing.                                            -->
  <recv request="INVITE" crlf="true" rrs="true">
  </recv>
  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurrences are concatenated (CRLF separated)       -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->
  <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>
  <send retrans="500">
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      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 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>
  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true">
  </recv>
  <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>
  <!-- Keep the call open for a while in case the 200 is lost to be     -->
  <!-- able to retransmit it if we receive the BYE again.               -->
  <timewait milliseconds="4000"/>
  <!-- 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>

uas_send_bye.xml 实现 SIPp 主动发送 BYE

默认 uas.xml 中，SIPp 作为 UAS 被动接收主叫发送的 BYE 消息。若需让 SIPp 主动发起挂断（即主动发送 BYE 消息），可创建uas_send_bye.xml 配置文件，核心逻辑是：接收 ACK 消息后，提取呼叫信息，主动构造并发送 BYE 消息。

核心修改说明

相较于适配代理的 uas.xml，uas_send_bye.xml 主要增加/修改了以下部分：

接收 ACK 消息时，通过 `ereg` 正则提取 To 头域（含 tag）和 From 头域信息，分别赋值给 `us` 和 `them` 变量；
添加 `<pause/>` 标签（默认暂停1秒，可通过 `milliseconds` 属性指定时长），模拟通话过程；
新增发送 BYE 消息的配置，通过 `start_txn="bye"` 标记 BYE 事务，使用提取的 `
新增接收 BYE 响应（200 OK）的配置，通过 `response_txn="bye"` 关联 BYE 事务。

uas_send_bye.xml 完整内容:

<?xml version="1.0" encoding="UTF-8" ?>
<!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             -->
<!--                                                                    -->
<!--                 Sipp default 'uas' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic UAS responder">
  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <!-- Adding ignoresdp="true" here would ignore the SDP data: that     -->
  <!-- can be useful if you want to reject reINVITEs and keep the       -->
  <!-- media stream flowing.                                            -->
  <recv request="INVITE" crlf="true" rrs="true">
  </recv>
  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurrences are concatenated (CRLF separated)       -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->
  <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>
  <send retrans="500">
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Record-Route:]
      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 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>
  <recv request="ACK" rrs="true">
    <action>
      <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
      <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
    </action>
  </recv>
  <Reference variables="_"/>
  <pause/>
  <send start_txn="bye" retrans="500">
    <![CDATA[
      BYE [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      From: [$us]
      To: [$them]
      [last_Call-ID:]
      CSeq: [cseq] BYE
      Contact: sip:[service]@[local_ip]:[local_port]
      Content-Length: 0
    ]]>
  </send>
  <recv response="200" response_txn="bye"/>
  <!-- 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>

关键说明：