[Freeswitch-users] in-dialog SIP options response with 0 RTP port

[Kashif Abbasi]

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 at 192.168.148.111:5080 SIP/2.0
Max-Forwards: 68
To: <sip:110237905476 at 192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr
From: "+99923468064011" <sip:+99923468064011 at 192.168.150.117>;tag=3814696699-708259
Call-ID: 215272-3814696699-708253 at 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 at 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 at 192.168.150.117>;tag=3814696699-708259
To: <sip:110237905476 at 192.168.148.111:5080;user=phone>;tag=H2rHgHZpgepHr
Call-ID: 215272-3814696699-708253 at ti-SBC-01.ti.com
CSeq: 2 OPTIONS
Contact: <sip:110237905476 at 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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