*From:* Bogdan-Andrei Iancu
*Sent:* Thursday, 21 March 2024 17:30
*To:* Stefan Carlsson; users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] SDP
Hi,
What OpenSIPS version are you using?
Regards,
Bogdan-Andrei Iancu
OpenSIPS
3.4
Sent from Outlook for Android<https://aka.ms/AAb9ysg>
From: Users on behalf of Bogdan-Andrei Iancu
Sent: Thursday, March 21, 2024 9:59:51 PM
To: Stefan Carlsson ; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP
Hi,
What OpenSIPS versi
p: a=sendrecv
replace_body("a=sendrecv", "a=sendonly"); # Indicate call hold
according RFC3264
// Regards …
Stefan
*From:* Bogdan-Andrei Iancu
*Sent:* Wednesday, 20 March, 2024 14:44
*To:* OpenSIPS users mailling list ; Stefan
Carlsson
*Subject:* Re: [OpenS
Regards …
_
Stefan Carlsson
*From:* Users *On Behalf Of
*Prathibha B
*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list
*Subject:* [OpenSIPS-Users] SDP
How to see the SDP in opensips?
--
Regards,
B.Prathibha
The online Manual should be your Bible here ;)
https://opensips.org/Documentation/Script-CoreVar-3-4#rb
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 14.03.2024 06:16, Prathibha B wrote:
How to see the SDP in open
impossible ☹
Regards …
_
Stefan Carlsson
From: Users On Behalf Of Prathibha B
Sent: Thursday, 14 March, 2024 05:16
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] SDP
How to see the SDP in opensips?
--
Regards,
B.Prathibha
How to see the SDP in opensips?
--
Regards,
B.Prathibha
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Hi Igor,
You are absolutely right. For anybody reading this in future, I figured it
works from branch_route because as Igor explained, that's where you have
access to unaltered packet. Thanks!
On Mon, Feb 12, 2024 at 10:35 AM Ihor Olkhovskyi
wrote:
> Usually you work in failure_route with packet
Usually you work in failure_route with packet that was in a state when you
called t_relay.
Best practice would be to work with rtp-related procedures in branch
routes, so you will get in failure_rroute unaltered packet, before
rtp_offer manipulations from previous time.
Cheers,
Ihor
Le sam. 10 f
Hi list,
When using rtpproxy_offer/answer, how can I rewrite SDP media IP/port if
for example the first route rejects the calls and I have to send the call
(in failure_route) to the next destination (where second destination uses
different media ip/port/rtpproxy set)?
If I just call rtpproxy_offer
ndrei Iancu" mailto:bog...@opensips.org>>
To: "OpenSIPS users mailling list" mailto:users@lists.opensips.org>>, "Pat Burke"
mailto:p...@voxtelesys.com>>
Date: 03/05/19 09:45
Subject: Re: [OpenSIPS-Users] SDP manipulation & rtpengine
Hi Pat,
...@voxtelesys.com
1801 23rd Avenue North | Suite 217 | Fargo, North Dakota 58102
-Original Message-
From: "Bogdan-Andrei Iancu"
To: "OpenSIPS users mailling list" , "Pat Burke"
Date: 03/05/19 09:45
Subject: Re: [OpenSIPS-Users] SDP manipulation & r
Hi Pat,
What you can do is to grab the SDP from the msg into a variable, to do
whatever fixes/change you have to directly in the variable and push the
body via variable to rtpengine.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Su
Hello:
I am using trying to manipulate parts of the SDP body before calling
rtpengine_offer / rtpengine_answer. However, any changes made via textops
functions such as subst_body, replace_body, replace_body_all, etc. do not seem
to impact the SDP that is sent to rtpengine.
In my particular
s@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP and ptime, maxptime removal
Hi, Stefan!
There is no bug here - ptime and maxptime are not associated to a codec, but to
the entire media session. So it should not be removed if you delete a certain.
Best regards,
Razvan
On 10/26/18 3:00 PM, S
Hi, Stefan!
There is no bug here - ptime and maxptime are not associated to a codec,
but to the entire media session. So it should not be removed if you
delete a certain.
Best regards,
Razvan
On 10/26/18 3:00 PM, Stefan Carlsson wrote:
Hi !
We have a serious issue that force us to remove al
Hi !
We have a serious issue that force us to remove all codecs except PCMA. We
use: codec_delete_except_re("PCMA"); from the sipmsgops module
but the function doesn't removes the associated ptime, maxptime. How can we
also remove this, any ideas or is it a bug in the sipmsgops module?
if this is, in fact, their
issue with the exchange.
Thanks,
Ben
*From: *Users on behalf of Adrian
Fretwell
*Reply-To: *OpenSIPS users mailling list
*Date: *Wednesday, January 24, 2018 at 2:06 PM
*To: *OpenSIPS users mailling list
*Subject: *[OpenSIPS-Users] SDP version increment without
: Users on behalf of Adrian Fretwell
Reply-To: OpenSIPS users mailling list
Date: Wednesday, January 24, 2018 at 2:06 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] SDP version increment without a change in SDP
Hello All, I have an issue for which I have not been able to find a
Hello All, I have an issue for which I have not been able to find a
definitive answer in the RFCs.
I use RTPProxy as part of NAT traversal so RTP streams are set up
between upstream provider and my proxy, and between my proxy and the UAC.
The problem I have, is UAC changes it's RTP port betw
Hi,
If you have an re-INVITE with active media, you just have to re-insert
the rtpengine, exactly as you did it for the initial INVITE. This will
properly handle the on-hold resume.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Su
Hello Bogdan-Andrei,
The issue with on hold RTP stream resume. I am not sure how to opensips
should handle properly.
I code bellow provide partial solution.
volga629
On Tue, 16 May, 2017 at 4:38 AM, Bogdan-Andrei Iancu
wrote:
Hello Volga,
What exactly does not work for you ? the detection
Hello Volga,
What exactly does not work for you ? the detection at SIP level of the
hold resume ? the actual RTP resume ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
http://www.opensips.org/events
Hello Everyone,
Thank you
That extract ip correctly.
$var(cline-ip) = $(rb{sdp.line,c}{s.select,2, });
My issue that I am trying make opensips handle music on hold resume
working properly.
I am not sure if possible do simpler way.
Relevant code
route[ONHOLD] {
if(is_method("INVITE|U
Hi Volga,
You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
http://www.opensips.org/events/Summit-2017Amster
Hi, Volga!
Check the SDP transformations[1].
[1] http://www.opensips.org/Documentation/Script-Tran-2-3#toc80
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:
Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN
Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.
volga629
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Anybody have any ideas?
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Thursday, March 24, 2016 3:16 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog
t: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
1.11.5
Thanks Liviu,
I will recompile and give it a try.
Louis
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016
Thanks Liviu,
I will recompile and give it a try.
Louis
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP
Louis Rochon
Sent: Wednesday, March 16, 2016 10:02 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5
SDP in ACK lost in OpenSIPS 1.11.5
This is something that used to work in OpenSIPS 1.8.1, but seems to have been
broken 1.11.5.
Using the
Anybody?
Louis
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Wednesday, March 16, 2016 10:02 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
SDP in ACK lost in OpenSIPS 1.11.5
This is something that used to work in OpenSIPS 1.8.1, but seems to have been
broken 1.11.5.
Using the B2BUA facilities, I make the second leg of the call using
b2b_init_request. Then, if required, I move the call to another user agent via
a bash shell script
Hi Riko,
I understand, but my knowledge over rtpengine is not so good. Maybe you
should ask over the rptengine project.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.01.2016 08:24, riko nir wrote:
Hi Andrei,
The opensips and rtpeng
When OpenSIPS handles the SIP messages carrying SDP (typically INVITE
request and 200 OK reply), OpenSIPS will communicate with rtpengine and
update the SDP with the new IP and port. Never tested, by AFAIK
rtpengine may know to handle SRTP...not sure, you may check with the
project.
Regards,
Ok. Incase if the media needs to go via rtpengine, then how the signaling
happens, and how the rtpengine is forwarding the media to the other end?
In case of DTLS, only the DTLS handlshake needs to be taken care by the
RtpEngine, and the SRTP is handled by some other media server, and only the
DTL
Hi Riko,
There are no SDP answers created by opensips - it simply changes the
received SDP offer and it forwards it to the next destination -
basically it is doing proxying (performing some changes too) of the SDP
offers between the end points.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Fou
Hi all,
How is the communication flow happens between opensips and rtpengine incase
of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip client, opensips is
creating sdp answer or rtpengine is creating sdp anwer or rtpengine just
updating ice information on
Checkout the textops module:
http://www.opensips.org/html/docs/modules/devel/textops.html
On Jun 19, 2014, at 10:47 PM, Ionut Muntean wrote:
> Hello,
>
> Is there a easy way to remove some attributes present in the SDP (b=RS:xx
> and/or b=RR:xx)?
>
> Thank you.
>
> / Ionut Muntean
>
> _
Hello,
Is there a easy way to remove some attributes present in the SDP
(b=RS:xx and/or b=RR:xx)?
Thank you.
/ Ionut Muntean
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Hi, Chen-Che!
No, OpenSIPS does not have any built-in mechanism for this. As far as I
understand, you sometimes need to rewrite the IP OpenSIPS advertises in
the SDP. You can specify the new IP in the second parameter of the
rtpproxy_offer/answer() function.
Best regards,
Razvan Crainea
Ope
Dear all,
I'm encountering an issue as follows. In the system, there are one SIP
server (OpenSIPS) and multiple RTP proxies. Each RTP proxy serves for one
region and the users in that region will use the corresponding RTP proxy for
media streaming relay. To achieve this, I make the users in the sa
Hi,
On Apr 16, 2012, at 10:34 AM, goup2010 wrote:
> Hello,
>
> I test opensips 1.8.
>
> When I use rtpproxy_offer() and rtpproxy_answer() in SDP , line
> appears two times.
>
> Is this correct?
>
Yes. The one on the stream takes precedence. As per RFC4566, sec5.7:
A session description MU
Hello,
I test opensips 1.8.
When I use rtpproxy_offer() and rtpproxy_answer() in SDP , line
appears two times.
Is this correct?
Here is SDP
v=0
o=- 3543552561 3543552561 IN IP4 192.168.1.102
s=pjmedia
c=IN IP4 192.168.1.102
b=AS:84
t=0 0
a=X-nat:0
m=audio 55946 RTP/AVP 8 0 101
c=IN IP4 88.99
Hello,
Thank you very much Bogdan and Ovidiu for quick response.
I have used dialog 'dlg_list_ctx' mi command. And I can inspect the dialog
as well as the associated qos session.
-Urmi
On Tue, Sep 1, 2009 at 7:49 PM, Ovidiu Sas wrote:
> Hello Urmi,
>
> Use the dialog 'dlg_list_ctx' mi comman
Hello Urmi,
Use the dialog 'dlg_list_ctx' mi command to inspect the internals of
the dialog along with the
associated qos session:
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#id272808
This is stated in the qos module doc:
http://www.opensips.org/html/docs/modules/1.5.x/qos.html#id2
Hi Urmi,
There are some functions in textops module to allow you codec inspection
and manipulation:
http://www.opensips.org/Main/News0034
Regards,
Bogdan
urmi lakkad wrote:
>
> Hello,
>
> I am using Opensips-1.5.1 with Asterisk.
> I want to test the QOS module functionality. I have configu
Hello,
I am using Opensips-1.5.1 with Asterisk.
I want to test the QOS module functionality. I have configured the dialog
module and its working fine.
Even I have set the flag of QOS module. Now my question is how to inspect
the SDP session.
Thanks for your attention.
-Urmi
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