rtp abnormal

[freeze-l]

this is my command to execute:
sipp 192.168.1.181:5060 -i 192.168.1.177 -p 5061 -sf call.xml -inf tt.csv -mi 192.168.1.177 -mp 7000 -m 1 -trace_msg -trace_err
181 is my freeswitch server,and 177 is my sipp server

tt.csv:SEQUENTIAL 8000;Abcd123.;1002;7000;7001

call.xml:

;tag=[pid][branch] To: Contact: Call-ID: [call_id] CSeq: 1 INVITE Session-Expires: 3600 Content-Type: application/sdp Content-Length: [len]

        v=0
        o=[field0] 53655765 2353687637 IN IP4 [local_ip]
        s=SIP Call
        c=IN IP4 192.168.1.177
        t=0 0
        m=audio [field3] RTP/AVP 8 0 101
        a=rtcp:[field4] IN IP4 [local_ip]
        a=sendrecv
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
    ]]>
</send>

<!-- 处理 407 Proxy Authentication Required -->
<recv response="407" auth="true"/>

<send>
    <![CDATA[
        ACK sip:[field2]@[remote_ip] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch-2]
        From: <sip:[field0]@[remote_ip]>;tag=[pid][branch-2]
        To: <sip:[field2]@[remote_ip]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 1 ACK
        Content-Length: 0
    ]]>
</send>

<!-- 重新发送 INVITE，带认证信息 -->
<send>
    <![CDATA[
        INVITE sip:[field2]@[remote_ip] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
        From: <sip:[field0]@[remote_ip]>;tag=[pid][branch]
        To: <sip:[field2]@[remote_ip]>
        Contact: <sip:[field0]@[local_ip]:[local_port];transport=udp>
        Call-ID: [call_id]
        CSeq: 2 INVITE
        [authentication username="[field0]" password="[field1]"]
        Session-Expires: 3600
        Content-Type: application/sdp
        Content-Length: [len]
        
        v=0
        o=[field0] 53655765 2353687637 IN IP4 [local_ip]
        s=SIP Call
        c=IN IP4 192.168.1.177
        t=0 0
        m=audio [field3] RTP/AVP 8 0 101
        a=rtcp:[field4] IN IP4 [local_ip]
        a=sendrecv
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
    ]]>
</send>

<!-- 继续 SIP 流程 -->
<recv response="100"/>
<recv response="200"/>

<send>
    <![CDATA[
        ACK sip:[field2]@[remote_ip] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch-3]
        From: <sip:[field0]@[remote_ip]>;tag=[pid][branch-3]
        To: <sip:[field2]@[remote_ip]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 2 ACK
        Content-Length: 0
    ]]>
</send>

<nop>
    <action>
        <exec play_pcap_audio="g711a.pcap"/>
    </action>
    <recv rtp_audio="true"/>
</nop>

<pause milliseconds="10000"/>

<send>
    <![CDATA[
        BYE sip:[field2]@[remote_ip] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-6]
        From: <sip:[field0]@[remote_ip]>;tag=[pid][branch-6]
        To: <sip:[field2]@[remote_ip]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 3 BYE
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        Max-Forwards: 70
        Content-Length: 0
    ]]>
</send>

<recv response="200" optional="true"/>

Now SIP signaling is normal, but my packet capture RTP shows that the port for the RTP packet sent from 177 to 181 is 0 like this Internet Protocol Version 4, Src: 192.168.1.177, Dst: 192.168.1.181 User Datagram Protocol, Src Port: 0, Dst Port: 31268 Real-Time Transport Protocol [Stream setup by SDP (frame 4)] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: ITU-T G.711 PCMA (8) Sequence number: 59141 [Extended sequence number: 59141] Timestamp: 2160 [Extended timestamp: 4294969456] Synchronization Source identifier: 0xdee0ee8f (3739283087) Payload […]:

I don't know where the problem lies