Hello,

i'm facing a strange issue on a trunk (on Internet). Here is the summary of the call flow :

1. The call is established normally (RTP Ports, Codec). Everything is fine.

2. During the call, Freeswitch receives a SIP OPTIONS with the SAME callid as the ongoing call.

3. Freeswitch answers 200 OK, but with a SDP with a 0 RTP port.

4. Our facing SIP Proxy uses 0 for next RTP trafic and then the communication continues with no sound

Is this a normal behaviour from Freeswitch ?

Do I have a possible configuration item to tune for that ?

Here is the SIP OPTIONS part of the SIP trace :

   ------------------------------------------------------------------------

recv 436 bytes from udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.284455:

   ------------------------------------------------------------------------

   OPTIONS sip:+3310129144152000@10.9.16.171:5060;transport=udp SIP/2.0

   Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0

   To: <sip:+3310129144152000@domain.net;user=phone>;tag=2tHZ5Xgv3HB8c

   From: <sip:+33624431415@domain.net;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000

   CSeq: 2 OPTIONS

   Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000

   Max-Forwards: 70

   Content-Length: 0

   User-Agent: Keyyo-S5-P

   ------------------------------------------------------------------------

send 905 bytes to udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.289599:

   ------------------------------------------------------------------------

   SIP/2.0 200 OK

   Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0

   From: <sip:+33624431415@domain.net;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000

   To: <sip:+3310129144152000@domain.net;user=phone>;tag=2tHZ5Xgv3HB8c

   Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000

   CSeq: 2 OPTIONS

   Contact: <sip:+3310129144152000@FREESWITCH_PUBLIC_IP:5060;transport=udp>

   User-Agent: FreeSWITCH

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY

   Supported: timer, path, replaces

   Allow-Events: talk, hold, conference, refer

   Content-Type: application/sdp

   Content-Length: 256

   v=0

   o=FreeSWITCH 1476957466 1476957467 IN IP4 FREESWITCH_PUBLIC_IP

   s=FreeSWITCH

   c=IN IP4 FREESWITCH_PUBLIC_IP

   t=0 0

   m=audio 0 RTP/AVP 8 101

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

   a=rtcp:19369 IN IP4 FREESWITCH_PUBLIC_IP

   ------------------------------------------------------------------------  

Dominique Jeannerod

Interact-iv.com

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