Hi Saul,
OK, aside the TCP part (which anyhow is scheduled for fixing) and some
extra parsing, does supporting WebRTC imply something more on the
OpenSIPS side ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/29/2012 12:42 PM, Saúl
Hi,
The names of the AVPS are wrong. Instead of:
/check_user_blacklist( $avp($fu), $avp($fd) ,$avp(rU) ) /
Try:
/check_user_blacklist( $avp(fu), $avp(fd) ,$avp(rU) ) /
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/26/2012 04:31 PM,
On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote:
Hi Saul,
OK, aside the TCP part (which anyhow is scheduled for fixing) and some extra
parsing, does supporting WebRTC imply something more on the OpenSIPS side ?
It requires that OpenSIPS is able to use SIP over a WebSocket
Hi Bobby,
I would say the leaking is related to RTPproxy stuff:
32106 msg_callbacks.c: msg_callback_add(51)
16053 rtpproxy_callbacks.c: rtpproxy_pre_fwd(51)
16053 rtpproxy.c: force_rtp_proxy(2848)
16053 rtpproxy.c: pkg_strdup(2301)
Have you enabled the rtpproxy_autobridge option in
Hi Bogdan,
Saul is correct. The key thing is to support WebSocket transport. WebRTC is
becoming quite popular and seems to be the thing of future. It is already
added in asterisk (version 11 released last week) and it is being added to
Kamalio. Using OverSIP and the OpenSIPS would make things
Hi Nick, you SOLVED my issue!!!
Thank you very much.
So, seems rtpproxy's bug is not resolved yet!!
Hope this long treat can help someone else.
regards
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp7581935p7582656.html
The latest improvement on Redis module are very interesting, thanks for the
contributiom, I would like to understand how the hiredis client is working
with a redis cluster
Where can I find information on the server groups/cluster support ?
Is it working with Redis Sentinel (the official high
Hi guys,
Thanks for this - I will take a look at this websocket to see what about
- is there any RFC or similar ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/31/2012 03:53 PM, Ali Pey wrote:
Hi Bogdan,
Saul is correct. The key thing
Hi Bogdan,
On Oct 31, 2012, at 5:05 PM, Bogdan-Andrei Iancu wrote:
Hi guys,
Thanks for this - I will take a look at this websocket to see what about - is
there any RFC or similar ?
Here you go: http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/
Regards,
--
Saúl Ibarra
Hello,
The current OpenSIPS cachedb_redis implementation does not support the
Redis Sentinel.
I see it's a relatively new concept, currently only part of the Redis
/unstable/ branch.
The way the cachedb_redis module currently handles clusters in OpenSIPS
is that, at startup, it takes a
Asterisk 11 has some early support for SIP over websockets, but that's far from
being compatible with WebRTC. The standards for WebRTC are still evolving and
require much more. It's a good step forward, but the ASterisk team is not there
yet... :-)
SIP over websockets is currently a draft that
2012/10/31 Ali Pey ali...@gmail.com
Using OverSIP and the OpenSIPS would make things just more complex
specially for larger deployments.
Really? IMHO it makes things MUCH MORE simple. Use OverSIP as an Outbound
EDGE Proxy and you are done. Otherwise wait for a proper TCP/TLS and
WebSocket
Which one sounds simpler? Having a new layer of proxies and extra hardware
on different software packages with their own set of configurations,
limitations and bugs than having WebSocket enabled on opensips and control
your routing logic all in one place off of same DB.
Regards,
Ali Pey
On Wed,
2012/10/31 Ali Pey ali...@gmail.com
Which one sounds simpler? Having a new layer of proxies and extra hardware
on different software packages with their own set of configurations,
limitations and bugs than having WebSocket enabled on opensips and control
your routing logic all in one place
31 okt 2012 kl. 18:17 skrev Ali Pey ali...@gmail.com:
Which one sounds simpler? Having a new layer of proxies and extra hardware on
different software packages with their own set of configurations, limitations
and bugs than having WebSocket enabled on opensips and control your routing
On 29 Oct 2012, at 12:11, Bogdan-Andrei Iancu wrote:
Hi Saul,
We were thinking at re-INVITE pinging from OpenSIPs level towards the caller
and callee. There will be 2 modes (at least this is the current plan).
1) remember all the time the last SDPs from each side and re-use them when
On Wed, Oct 31, 2012 at 3:03 PM, Dan Pascu d...@ag-projects.com wrote:
On 29 Oct 2012, at 12:11, Bogdan-Andrei Iancu wrote:
Hi Saul,
We were thinking at re-INVITE pinging from OpenSIPs level towards the caller
and callee. There will be 2 modes (at least this is the current plan).
1)
Hi Bogdan:
Yes, we are using autobridge. Can I email you/link you the script privately?
Thanks for the help,
BobbyS
Sent from my iPhone
On Oct 31, 2012, at 8:37 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote:
Hi Bobby,
I would say the leaking is related to RTPproxy stuff:
32106
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