I encountered a situation where a websocket connection was getting
killed (presumably by the user's router) after ~ 2-3 minutes of inactivity.
To remedy this I looked into the keepalive option which was introduced
in Asterisk 11, as it is both simple and supported by libraries such as
JsSIP.
Hi Jeremy,
Another option is just to activate TCP keepalive on your Linux server.
Take a look at :
http://tldp.org/HOWTO/TCP-Keepalive-HOWTO/usingkeepalive.html
This what we use to maintain SIP sessions over TCP that sometimes get
killed by NAT routers.
Hope that helps,
Philippe
2014-02-11
Another option is to use a SIP proxy
My typical scenario involves repro talking to the WS or WSS client and
then using TCP to talk to Asterisk or whatever else
repro supports the CRLF keepalive mechanism over all streams including
TCP, TLS, WS and WSS.
Is is also possible to just set a very
Thanks for the feedback, it's good to have alternatives. FYI, as a workaround I
am
currently using qualify=yes.
However I'd still like to understand why the keepalive option is currently
not being
applied to websockets. The option is there, and is implemented for multiple
transports, so
I'd
On Tue, Feb 11, 2014 at 1:40 PM, Jeremy Lainé jeremy.la...@m4x.org wrote:
Thanks for the feedback, it's good to have alternatives. FYI, as a workaround
I am
currently using qualify=yes.
However I'd still like to understand why the keepalive option is currently
not being
applied to
On 11 Feb 2014 21:21, Matthew Jordan mjor...@digium.com wrote:
Either this is by design, in which case it needs to be documented, or it's
an oversight,
in which case I'd be happy to fix it.
I'd go with oversight.
Thanks for your answer I will propose a patch in that case.
I have a
On 14-02-11 05:42 PM, Jeremy Lainé wrote:
On 11 Feb 2014 21:21, Matthew Jordan mjor...@digium.com wrote:
Either this is by design, in which case it needs to be documented, or it's
an oversight,
in which case I'd be happy to fix it.
I'd go with oversight.
Thanks for your answer I
