Hi opensips users,
I have a couple of questions and I'm hoping someone can point me in the right
direction.
We have a SIP PBX that doesn't do WSS for example using sipml5. I'm trying to
put opensips in the middle of the SIP PBX and the WSS client with limited
success using mid-registrar in ope
All,
Never mind it is registering now. I still have some work to do but after more
investigation the SIP server was deleting the second VIA which was the via for
WSS. Now on the register I save it to an avp and then in my register's
onreply_route I just do an append_hf("Via: ..."). The client s
Hi list,
This might not be the correct list for this but maybe someone might be able to
point me in the correct direction. I'm trying to use opensips as a webrtc
gateway. It mostly works I'm able to call a legacy sip phone connected to my
SIP server. The reason why it only mostly works is I hav
st regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/23/2018 05:01 PM, Esty, Ryan wrote:
> Hi list,
>
> This might not be the correct list for this but maybe someone might be
> able to point me in the correct direction. I’m trying to use opensips
>
ur devices. The
packetization-mode was being set to one in the Janus example also.
Ryan Esty
-Original Message-----
From: Esty, Ryan
Sent: Tuesday, April 3, 2018 1:28 PM
To: users@lists.opensips.org
Subject: RE: [OpenSIPS-Users] h264 webrtc and opensips
Razvan,
Thanks for getting back to
Hi opensips list,
First some background I'm trying to use opensips as a webrtc proxy. I found out
that the packets for the invite going to my sip server are too big for my sip
server. It doesn't like packets to be over 4000 bytes. I'm trying to take what
I can out of the sip packets like codes
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 04/09/2018 05:22 PM, Esty, Ryan wrote:
Hi opensips list,
First some background I'm trying to use opensips as a webrtc proxy. I found out
that the packets for the invite going to my sip server are too big for my sip
s
ted to OpenSIPS, rather than
rtpengine, since you are getting the error in OpenSIPS, is that right?
Can you confirm what version of OpenSIPS you are using?
Best regards,
Răzvan
On 04/13/2018 03:42 PM, Esty, Ryan wrote:
> Bogdan-Andrei,
>
> Thanks for the information just in case som
t 2.3 nightly has this fixed,
so I'd suggest you to use that opensips version.
Best regards,
Răzvan
On 04/13/2018 04:31 PM, Esty, Ryan wrote:
> Razvan,
>
> Rtpengine is printing out this: rtpengine:parse_flags: error processing flag
> `codec-strip-VP8': unknown error. As I look a
rsion.
Best regards,
Răzvan
On 04/13/2018 04:31 PM, Esty, Ryan wrote:
> Razvan,
>
> Rtpengine is printing out this: rtpengine:parse_flags: error processing flag
> `codec-strip-VP8': unknown error. As I look at it now I don't see how it
> could be your issue. You aren&
Hi list,
I'm hoping someone can direct me to some documentation or help me out. I'm
trying to send an invite to something outside my domain. Everything seems to
work but no packets come back to opensips. I see the packets on the wire coming
to the machine but it looks like the tcp port that was
List,
I was able to get around my problem by disabling tcp_async. I was seeing a
whole bunch of Polling is overdue. This doesn't seem like the right solution
though, more like I fixed it by some timing side effect.
Ryan
From: Esty, Ryan
Sent: Wednesday, April 18, 2018 8:59 AM
To: &
Bogdan-Andrei,
I was going to look into this a little more before reporting it but I'm having
issues trying to get rtcp-mux-require to be sent down to rtpengine. From what I
can tell I need this to work with chrome's webrtc. Since this is crunch time I
didn't want a potential bug not being cove
our ticketing system [1] to keep
track of this issue as well.
[1] https://github.com/OpenSIPS/opensips/issues
Best regards,
Răzvan
On 05/17/2018 03:27 PM, Esty, Ryan wrote:
> Bogdan-Andrei,
>
> I was going to look into this a little more before reporting it but I'm
> having
14 matches
Mail list logo
