[Freeswitch-users] Answer to SIP OPTIONS with RTP Port 0 in SDP

[Dominique Jeannerod]

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 at 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 at domain.net;user=phone>;tag=2tHZ5Xgv3HB8c
   From: <sip:+33624431415 at 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 at domain.net
;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000
   To: <sip:+3310129144152000 at domain.net;user=phone>;tag=2tHZ5Xgv3HB8c
   Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000
   CSeq: 2 OPTIONS
   Contact: <sip:+3310129144152000 at 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*
Mobile: +33 698 778 763

Headquarter : 600 rue Felix Trombe – tecnosud – 66100 Perpignan - France
France : 0826 10 23 21 (0,15€/mn) International +33 (0)4 20 10 45 45
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161021/e1b23286/attachment.html