rainea
<raz...@opensips.org>
Sent: Monday, April 16, 2018 2:41 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi, Ryan!
That error is actually triggered by opensips, rtpengine module. There
was a bug related to this issue that was fixed on 21st
version of opensips is 2.3.3.
Ryan
-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan
Crainea
Sent: Friday, April 13, 2018 9:17 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi, Ryan!
I think the issue
-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan
Crainea
Sent: Friday, April 13, 2018 9:17 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi, Ryan!
I think the issue you are talking about is related to OpenSIPS
ers@lists.opensips.org>; Esty,
Ryan <ryan.e...@necect.com>
*Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi Ryan,
yeah, this happens because OpenSIPS applies all the changes at the end,
when the message is about to be sent out. As a side effect, when sending
the SDP to rtpengi
ers@lists.opensips.org>; Esty, Ryan
<ryan.e...@necect.com>
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi Ryan,
yeah, this happens because OpenSIPS applies all the changes at the end, when
the message is about to be sent out. As a side effect, when sending the SDP t
Hi Ryan,
yeah, this happens because OpenSIPS applies all the changes at the end,
when the message is about to be sent out. As a side effect, when sending
the SDP to rtpengine, opensips does not see its own previous changes
over the body (changes are still pending).
Usually there are easy
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
Wrong mailing list? ;-)
On Aug 9, 2012, at 6:06 AM, Joe Flemmings wrote:
I’m using Inbound “Mod event socket” and want to associate the codec event
with a call in the system.
When the event is first fired by FreeSwitch, it has a new Unique-ID and there
is no way to associate it with any
Yes, Sorry about that.
Joe.
On Fri, Aug 10, 2012 at 12:39 AM, Saúl Ibarra Corretgé s...@ag-projects.com
wrote:
Wrong mailing list? ;-)
On Aug 9, 2012, at 6:06 AM, Joe Flemmings wrote:
I’m using Inbound “Mod event socket” and want to associate the codec
event with a call in the system.
I’m using Inbound “Mod event socket” and want to associate the codec event
with a call in the system.
When the event is first fired by FreeSwitch, it has a new Unique-ID and
there is no way to associate it with any other channels.
How can i associate this event so i know what call it belongs to.
Hi everyone,
How can I do the translation of audio codec betwen the users ?
Example:
User1 - OPENSIPS
- User2
G711 G711 - G729 G729
Best regards.
Sam.
--
View this message in context:
Hi everyone,
How can I do the translation of audio codec betwen the users ?
Example:
User1 OPENSIPS --- User2
G711 G711 - G729 G729
Best regards.
Sam.
--
View this message in context:
If all you have is user one, user two, and opensips, you will never have to
worry about this because the call will not set up unless the two endpoints can
settle on a codec they both support. No transcoding needed.
Richard
On May 20, 2010, at 6:06 AM, samoh wrote:
Hi everyone,
How can
Hi Richard,
Thank you for your reply.
that is just an example to explain my question.
Best regards.
Sam.
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Codec-translation-tp5078977p5081120.html
Sent from the OpenSIPS - Users mailing list archive at
How to configure Opensips (version 1.5.0) to use a particular CODEC say..
Speex.?
--
View this message in context: http://n2.nabble.com/CODEC-tp3465202p3465202.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing
smadhoo6 wrote:
How to configure Opensips (version 1.5.0) to use a particular CODEC say..
Speex.?
This is like asking how to put the milk back in the cow with JSON.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct :
Hi Bogdan,
Just curious if you had a chance to see what the status is of this SDP
manipulation code.
Thanks,
Jeff
On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
Hi Jeff,
I have some code from last year - let me check it out and see what is
the status.
Regards,
Hi Jeff,
This is Andrei. I am currently working on this right now. It should be finished
in a couple of days if nothing comes up.
Andrei.
--- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote:
Just curious if you had a chance to see what the status is
of this SDP
manipulation
Great, Andrei. I look forward to being able to test it.
- Jeff
On 7/22/09 8:59 AM, andrei dragus andreidra...@yahoo.com wrote:
Hi Jeff,
This is Andrei. I am currently working on this right now. It should be
finished in a couple of days if nothing comes up.
Andrei.
--- On Wed,
Hello Andrei,
Are you using the core sdp parser implementation?
Ovidiu
On Wed, Jul 22, 2009 at 8:59 AM, andrei dragusandreidra...@yahoo.com wrote:
Hi Jeff,
This is Andrei. I am currently working on this right now. It should be
finished in a couple of days if nothing comes up.
Andrei.
Yes.
--- On Wed, 7/22/09, Ovidiu Sas o...@voipembedded.com wrote:
From: Ovidiu Sas o...@voipembedded.com
Hello Andrei,
Are you using the core sdp parser implementation?
Ovidiu
On Wed, Jul 22, 2009 at 8:59 AM, andrei dragusandreidra...@yahoo.com
wrote:
Hi Jeff,
This is
Perfect. Let me know if you need more features from the sdp parser.
Thanks,
Ovidiu
On Wed, Jul 22, 2009 at 11:32 AM, andrei dragusandreidra...@yahoo.com wrote:
Yes.
--- On Wed, 7/22/09, Ovidiu Sas o...@voipembedded.com wrote:
From: Ovidiu Sas o...@voipembedded.com
Hello Andrei,
Are
Hi Jeff,
Andrei, the author of the memchached module is working on this right
now. I guess this will be ready soon ;)
Regards,
Bogdan
Jeff Pyle wrote:
Hi Bogdan,
Just curious if you had a chance to see what the status is of this SDP
manipulation code.
Thanks,
Jeff
On 6/15/09 6:36
Exactly. This would be so incredibly useful.
Imagine a customer with a SIP-based PRI and a T1¹s worth of bandwidth behind
it. This is a common scenario for my customers. We try to avoid running
all G.729 wherever possible because of the obvious and ugly audio quality
hit. At G.711 I can only
On Tuesday 16 June 2009 12:48:20 Jeff Pyle wrote:
Exactly. This would be so incredibly useful.
Imagine a customer with a SIP-based PRI and a T1¹s worth of bandwidth
behind it. This is a common scenario for my customers. We try to avoid
running all G.729 wherever possible because of the
Hi Raúl,
It comes down to the service we're offering. When a customer buys a PRI
from us, they don't care or even know in many cases it's driven by VoIP.
They're buying a PRI -- they expect PRI quality. It is my opinion that
G.729 does not provide that quality. But, I also do audio production,
Jeff, I hadn't thought of those scenarios.. I suppose with the dialog module
you can do some pretty neat stuff.. this is a great idea..
On Tue, Jun 16, 2009 at 9:03 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Raúl,
It comes down to the service we're offering. When a customer buys a PRI
Hi Bogdan,
It¹s been a little while since we talked about this. I was wondering if
there was anything in the works to detect and/or manipulate the codecs
present in an SDP.
- Jeff
On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote:
This idea is quite standard in SBCs,
Hi Jeff,
I have some code from last year - let me check it out and see what is
the status.
Regards,
Bogdan
Jeff Pyle wrote:
Hi Bogdan,
It’s been a little while since we talked about this. I was wondering
if there was anything in the works to detect and/or manipulate the
codecs present
my $0.02 here.. I'm not sure if this is what you guys had in mind.. but I've
had situations where this would be handy.. It'd need to have some way of
identifing the codec (by number?). I'm not sure if the core really has
anything that parses the SDP by RFC spec, so I'm not sure how you'd do
things
Hi Brett,
Brett Nemeroff wrote:
my $0.02 here.. I'm not sure if this is what you guys had in mind..
but I've had situations where this would be handy.. It'd need to have
some way of identifing the codec (by number?). I'm not sure if the
core really has anything that parses the SDP by RFC
Nice.. That's about what I had in mind.. I'll be looking for more discussion
regarding this. :)-Brett
On Mon, Jun 15, 2009 at 8:21 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Brett,
Brett Nemeroff wrote:
my $0.02 here.. I'm not sure if this is what you guys had in mind.. but
The core has an SDP parser which is able to parse and correlate codec
names with payload types. The only module using the sdp parser is the
qos.
The parsed sdp structure can be inspected via dialog mi commands:
http://lists.opensips.org/pipermail/devel/2008-December/001708.html
Regards,
Ovidiu
This idea is quite standard in SBCs, typically called codec profiles, where
you permit only certain codecs to be passed through the SBC in an INVITE and
all the rest are stripped out.
We use it to get around interop issues with certain codecs. E.g. we have
some end devices/customers that have
Hi Jeff,
right now there is only available some functionality to check the codecs
(to see what codecs are advertised in the SDP)... What exactly are you
looking for (like codec ops) ?
Regards,
Bogdan
Jeff Pyle wrote:
Bogdan,
Some months back you mentioned an upcoming feature that would
Hello Jeff,
Take a look also at the qos module. It keeps per dialog track of the
negotiated sdp.
There's an API that can be used by other modules sitting on top of the
qos module.
Regards,
Ovidiu Sas
On Thu, Jan 29, 2009 at 8:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Bogdan,
I'm
Bogdan,
Some months back you mentioned an upcoming feature that would allow Opensips
to manipulate the codecs present in the SDP. Just wondering if there is
anything available to test yet. This feature, in combination with dialog
contexts, will be of great use to us to allow us to take a guess
El Viernes, 28 de Noviembre de 2008, Tseveendorj Ochirlantuu escribió:
Hello,
What kind of codecs included in OpenSIPS? G.711mu, G711a ,G729 ...
Do you understant that OpenSIPS is a *SIP proxy* and *never* a media server or
media proxy?
OpenSIPS knows nothing about codec or RTP.
--
Iñaki
Hello,
What kind of codecs included in OpenSIPS? G.711mu, G711a ,G729 ...
Thank you
Sincerely,
Tseveen.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
For the clarity codecs are only in the end-points (e.g. SIP phone).
MediaProxy is not an end point and is codec agnostic. So it does not
need to support a particular codec, it relays UDP/RTP packets
regardless of what codec they have inside.
Adrian
On Nov 28, 2008, at 1:34 PM, Giuseppe
40 matches
Mail list logo
