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
shou
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 $
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:
Hello,
have you printed the message in syslog with $mb to see its content?
What is the content-lenght value sent by client and the one sent out by
kamailio.
Cheers,
Daniel
On 28/10/14 16:21, Ricardo Martinez wrote:
>
> Hello.
>
> I’m having some problems using websocket to communicate a webRTC
Hello.
I’m having some problems using websocket to communicate a webRTC client
with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket
I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length.
When I made the capture on the other leg (the pur
Hello.
I’m having some problems using websocket to communicate a webRTC client
with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket
I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length.
When I made the capture on the other leg (the pur
