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Hi, I am having an issue in freeswitch case where during a normal in progress audio call session, A party is sending in-dialog SIP OPTIONS message to freeswitch but somehow freeswitch is responding back with 200OK including the SDP and that SDP somehow contains
"0" media/RTP port (m=audio 0 RTP/AVP 0 101). This is causing one way muting during the session. Can someone please help how freeswitch can be configured to stop behaving like this? Thanks
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<div>>>> In-dialog SIP OPTIONS from source side to Freeswitch</div>
<div>OPTIONS sip:110237905476@192.168.148.111:5080 SIP/2.0</div>
<div>Max-Forwards: 68</div>
<div>To: <sip:110237905476@192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr</div>
<div>From: "+99923468064011" <sip:+99923468064011@192.168.150.117>;tag=3814696699-708259</div>
<div>Call-ID: 215272-3814696699-708253@ti-SBC-01.ti.com</div>
<div>CSeq: 2 OPTIONS</div>
<div>Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, UPDATE, MESSAGE, PUBLISH</div>
<div>Via: SIP/2.0/UDP 192.168.150.117:5060;branch=z9hG4bK0f514d4ecdb762b3b0f17a0d489897fd</div>
<div>Contact: <sip:+99923468064011@192.168.150.117:5060></div>
<div>Accept: application/sdp</div>
<div>Accept: application/isup</div>
<div>Accept: application/xml</div>
<div>Content-Length: 0</div>
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<div>>>> Reply from Freeswitch with SDP</div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 192.168.150.117:5060;branch=z9hG4bK0f514d4ecdb762b3b0f17a0d489897fd</div>
<div>From: "+99923468064011" <sip:+99923468064011@192.168.150.117>;tag=3814696699-708259</div>
<div>To: <sip:110237905476@192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr</div>
<div>Call-ID: 215272-3814696699-708253@ti-SBC-01.ti.com</div>
<div>CSeq: 2 OPTIONS</div>
<div>Contact: <sip:110237905476@192.168.148.111:5080;transport=udp></div>
<div>User-Agent: FreeSWITCH-mod_sofia/1.10.4-release-16-133fc2c870~64bit</div>
<div>Accept: application/sdp</div>
<div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div>
<div>Supported: timer, path, replaces</div>
<div>Allow-Events: talk, hold, conference, refer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 222</div>
<div>v=0</div>
<div>o=FreeSWITCH 1605678925 1605678926 IN IP4 192.168.148.111</div>
<div>s=FreeSWITCH</div>
<div>c=IN IP4 192.168.148.111</div>
<div>t=0 0</div>
<div><span style="background-color: rgb(255, 255, 0);">m=audio 0 RTP/AVP 0 101</span></div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
a=ptime:20<br>
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Regards</div>
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Kash</div>
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