From gascagonzalo@gmail.com Thu Oct 30 19:18:10 2014 From: Gonzalo Gasca To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] websocket and SIP Date: Thu, 30 Oct 2014 11:18:03 -0700 Message-ID: In-Reply-To: <5452187D.1070805@gmail.com> MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="===============1628245293==" --===============1628245293== Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable Hi Ricardo, I have a similar setup working: sipml5 -wss-> Kamailio -udp-> GW (FS) I use Freeswitch with UDP and works fine, as you can see initial Invite with SDP for Webrtc clients using sipMl5 is normally pretty big (audio+video) and normally if you are proxying that message the remote end should reassamble it, the best way to test it is to a tcpdump on the other side and see what the OS is receiving. On Thu, Oct 30, 2014 at 3:52 AM, Daniel-Constantin Mierla wrote: > Hello, > > the problem can be UDP fragmentation -- the gateway stack is not able to > handle UDP fragments. If the gateway supports tcp, then use this transport > layer. > > Cheers, > Daniel > > > On 29/10/14 21:42, Ricardo Martinez wrote: > > Hello Daniel. > > I have printed the $mb in the kamailio debug and the $ml : > > The SIP message in the client side has 2759 bytes. > > This is what I get from the kamailio at the entrance leg : > > Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: