Good morning everyone,
I finally figured out what was happening. It was related to asterisk configuration, there
was no problem with kamailio configuration.
When configuring the endpoint for kamailio in asterisk I set webrtc parameter to yes, this
parameter is a shortcut for settings all this parameters
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes
Instead of setting webrtc to yes on pjsip.conf I configured all the parameters that it
changes. After this everything seems to work properly.
Thank you all for your replies, it helped me to figure this out.
Also I wanted to ask if anyone knows if there’s a guide for this. I haven’t find out
anything appart from this article
(
https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). It’s very
helpful but it didn’t explain how to do the same but using websocket to connect extensions
to kamailio.
Thank you again for your time
Samuel Moya Tinoco
Departamento de Sistemas y Redes
Móvil: (+34) 606985997
smoya@vivelibre.es<mailto:smoya@vivelibre.es>
[cid:image001.png@01DA1618.E4BB7E90]
Soluciones inteligentes
para la autonomía personal
De: Social Boh via sr-users <sr-users(a)lists.kamailio.org>
Enviado el: viernes, 10 de noviembre de 2023 23:29
Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
CC: Social Boh <social(a)bohboh.info>
Asunto: [SR-Users] Re: Working with kamailio websocket and asterisk
You can capture signaling with sngrep and SIPTRACE module. If there is a UDP Fragmentation
the INVITE to de udp device don't receive a reply
https://www.voztovoice.org/?q=node/3020<https://linkprotect.cudasvc.com/…
Regards
---
I'm SoCIaL, MayBe
El 10/11/2023 a las 8:43 a. m., James Browne via sr-users escribió:
Hi
I've missed some of the email replies in this thread (I see them on
lists.kamailio.org<http://lists.kamailio.org> website.)
- It's been suggested that NAT may be the root of the problem. However, that can't
explain how the 200-OK for the CANCEL and then 100-Trying for the INVITE worked.
- Path and Record-Route headers were mentioned, but they're not relevant for receipt
of INVITE responses (and not relevant for this anyway because you've TOPOS
configured).
The size of the message (200-OK for the INVITE) is often significant. I note that the
INVITE was well over 1500 bytes, so I expect that the response may well have been large
too.
In my experience, when this sort of problem happens, there's often a problem with IP
fragmentation (a router not routing the fragments) or with a misconfigured MTU on the
network somewhere, quite possibly at the asterisk machine. This is almost certainly an IP
problem, not a SIP problem.
James
On Thu, 9 Nov 2023 at 19:46, Ihor Olkhovskyi via sr-users
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> wrote:
Samuel,
Have you used Path header?
As described here:
https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/<https…
Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> a écrit :
Good evening everyone,
I’m trying to set a scenario with my extensions registered against kamailio via wss and
then once authenticated forward the register to my asterisk.
After this, I use dispatcher module to balance invites to my asterisks.
I have a few questions, the first part seems to be working as I want but when asterisk
receives an invite and responds with INVITE (SDP) it doesn’t appear on my kamailio (but
every other message appears) You can see it in this sngrep captures
Kamailio:
[cid:image002.png@01DA1619.5D0D7F10]
Asterisk
[cid:image003.png@01DA1619.5D0D7F10]
I guess it is because something in my asterisk pjsip configuration is wrong but I can’t
figure out what. So I’m start to think that maybe It is because I need to configure an
rtpproxy app to change SDP and make RTP traffic go through kamailio
Maybe there’s something else I need to do, or there’s a guide to do this. But I can’t find
anything
Any help will be very appreciated
Thank you in advance
Samuel Moya Tinoco
Departamento de Sistemas y Redes
Móvil: (+34) 606985997
smoya@vivelibre.es<mailto:smoya@vivelibre.es>
[cid:image001.png@01DA1618.E4BB7E90]
Soluciones inteligentes
para la autonomía personal
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Best regards,
Ihor (Igor)
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