:12 PM
To: Daniel-Constantin Mierla ; Kamailio (SER) - Users
Mailing List
Subject: Re: [SR-Users] 488 Not Acceptable Here when remove codecs with rtcp-fb
Hello,
I can test the patch and give you feedback.
Best Regards,
Jose
On Mon, May 11, 2020 at 12:18 PM Daniel-Constantin Mierla
mailt
Hello,
I can test the patch and give you feedback.
Best Regards,
Jose
On Mon, May 11, 2020 at 12:18 PM Daniel-Constantin Mierla
wrote:
> Hello,
>
> sure, make a pull request for it.
>
> To be easy to merge, here are some guidelines for contributions
>
> *
>
> https://github.com/kamailio/ka
Hello,
sure, make a pull request for it.
To be easy to merge, here are some guidelines for contributions
*
https://github.com/kamailio/kamailio/blob/master/.github/CONTRIBUTING.md#contributing-code-or-content
Cheers,
Daniel
On 11.05.20 12:48, Federico Santulli wrote:
> I did a patch over the
I did a patch over the SDPOPS module to do this.
If needed I can push it over git.
Kind regards.
Federico Santulli
NHM - S.R.L.
Via Raffaello Sanzio, 88
81031 Aversa (CE)
Italy
> Il giorno 5 mag 2020, alle ore 15:55, José Lopes
> ha scritto:
>
> Hello,
>
> I am using KSR.sdpops.keep_codec
Hello,
I am sending in attach a pcap with a call between two webrtc clients, that
reproduces this scenario.
I am using as example the configuration at
https://github.com/havfo/WEBRTC-to-SIP .
I apply this change on the kamailio configuration to reproduce this
scenario:
@@ -350,6 +350,8 @@ request
Hello,
Sorry, I forgot to mention that, between the call of the two webrtc
clients, there is a B2BUA that only supports SIP UDP and RTP, so I need to
use rtpengine to translate from DTLS/SRTP to RTP.
I will try to make a call between the two webrtc clients and only use
kamailio without rtpengine t
This 488 think reminds me that SIP over webrtc is broken.
Webrtc UAS (at least JsSIP) cannot issue 488 before the UAS has started
to ring. It is very frustrating for the callee to get such a spam ring.
RFC3261 section "8.2 UAS Behavior" tells:
Note that request processing is atomic. If a re
José Lopes writes:
> Hello Daniel,
>
> Thanks for your reply.
>
> On the next link, I put the original SDP from webrtc client and the SDP
> after kamailio that exposes the issue.
> I am using Kamailio version 5.3.3 with sdpops and rtpengine module.
>
> https://pastebin.com/bYr0AcVT
Perhaps it i
Hello Daniel,
Thanks for your reply.
On the next link, I put the original SDP from webrtc client and the SDP
after kamailio that exposes the issue.
I am using Kamailio version 5.3.3 with sdpops and rtpengine module.
https://pastebin.com/bYr0AcVT
Best Regards,
Jose Lopes
On Tue, May 5, 2020 at
Hello,
can you give a sample SDP that exposes the issue so we can analyze with
the code and be able to test?
Cheers,
Daniel
On 05.05.20 15:55, José Lopes wrote:
> Hello,
>
> I am using KSR.sdpops.keep_codecs_by_name to remove some codecs.
> When I make a call between two webrtc clients and I kee
Hello,
I am using KSR.sdpops.keep_codecs_by_name to remove some codecs.
When I make a call between two webrtc clients and I keep only VP8 video
codec, the browser rejects the call with 488 Not Acceptable Here.
I notice the attributes a=rtcp-fb: of codecs removed are kept.
I removed that attribute
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