in-dialog SIP options response with 0 RTP port
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
In-dialog SIP OPTIONS from source side to Freeswitch OPTIONS sip:110237905476@192.168.148.111:5080 SIP/2.0 Max-Forwards: 68 To: <sip:110237905476@192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr From: "+99923468064011" <sip:+99923468064011@192.168.150.117>;tag=3814696699-708259 Call-ID: 215272-3814696699-708253@ti-SBC-01.ti.com CSeq: 2 OPTIONS Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, UPDATE, MESSAGE, PUBLISH Via: SIP/2.0/UDP 192.168.150.117:5060;branch=z9hG4bK0f514d4ecdb762b3b0f17a0d489897fd Contact: <sip:+99923468064011@192.168.150.117:5060> Accept: application/sdp Accept: application/isup Accept: application/xml Content-Length: 0
Reply from Freeswitch with SDP SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.117:5060;branch=z9hG4bK0f514d4ecdb762b3b0f17a0d489897fd From: "+99923468064011" <sip:+99923468064011@192.168.150.117>;tag=3814696699-708259 To: <sip:110237905476@192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr Call-ID: 215272-3814696699-708253@ti-SBC-01.ti.com CSeq: 2 OPTIONS Contact: <sip:110237905476@192.168.148.111:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.10.4-release-16-133fc2c870~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Length: 222 v=0 o=FreeSWITCH 1605678925 1605678926 IN IP4 192.168.148.111 s=FreeSWITCH c=IN IP4 192.168.148.111 t=0 0 m=audio 0 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Regards Kash
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Kashif Abbasi