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:image001.png@01DA0E88.5F82F510] Asterisk [cid:image002.png@01DA0E88.5F82F510]
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.esmailto:smoya@vivelibre.es
[cid:image003.png@01DA0E88.5F82F510]
Soluciones inteligentes para la autonomía personal
Samuel,
Have you used Path header?
As described here: https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/
Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users < 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:
Asterisk
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
Soluciones inteligentes para la autonomía personal
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi
I've missed some of the email replies in this thread (I see them on 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> wrote:
Samuel,
Have you used Path header?
As described here: https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/
Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users < 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:
Asterisk
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
Soluciones inteligentes para la autonomía personal
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
-- Best regards, Ihor (Igor) __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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
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 wrote:
Samuel, Have you used Path header? As described here: https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/ Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users <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: Asterisk 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> Soluciones inteligentes para la autonomía personal __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: -- Best regards, Ihor (Igor) __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email tosr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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.esmailto:smoya@vivelibre.es
[cid:image001.png@01DA1618.E4BB7E90]
Soluciones inteligentes para la autonomía personal
De: Social Boh via sr-users sr-users@lists.kamailio.org Enviado el: viernes, 10 de noviembre de 2023 23:29 Para: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org CC: Social Boh social@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/3020https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.voztovoice.org%2f%3fq%3dnode%2f3020&c=E,1,1DgEbz7X424LFKG4Ezfih6WwWCTP-dnl8OM0J4J5QN7SCIepwVMT1tBpk2KRr8r-geCBjklMmvkJ6IoKrTrNwT0ipk_XpmxSoiMHCDbXyBiI&typo=1
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.orghttp://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.orgmailto: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://linkprotect.cudasvc.com/url?a=https%3a%2f%2fblog.irontec.com%2fintegracion-kamailio-y-asterisk-con-path%2f&c=E,1,3hT4U2xUpcvyHLQb64rQ8vEeom4bT4nXSO5LfHuHc2YVi__hEg358hj8Grw-M0UYkKjzjkW2IfT7SjrYokjQdOhvJoCXLDnn1QEXyDoUlQ,,&typo=1
Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users <sr-users@lists.kamailio.orgmailto: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.esmailto:smoya@vivelibre.es
[cid:image001.png@01DA1618.E4BB7E90]
Soluciones inteligentes para la autonomía personal
__________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.orgmailto:sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
-- Best regards, Ihor (Igor) __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.orgmailto:sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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200 on Cancel is 1 Hop response (as CANCEL is 1 hop request ) as well as 100 Trying
Retransmission of 200 response not necessary shows problem of the response delivery and in most of cases it is about not receiving ACK ( whish is routed bases on Contact and routes set in 200 )
On Sat, 11 Nov 2023, 01:34 James Browne via sr-users, < sr-users@lists.kamailio.org> wrote:
Hi
I've missed some of the email replies in this thread (I see them on 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> wrote:
Samuel,
Have you used Path header?
As described here: https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/
Le mar. 7 nov. 2023 à 10:57, SAMUEL MOYA TINOCO via sr-users < 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:
Asterisk
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
Soluciones inteligentes para la autonomía personal
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
-- Best regards, Ihor (Igor) __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
You're not involving Proxy routing ( record-route/route. Read paragraph 16 of the rfc3261 and take a look into kamailio.cfg example provided after installation if the kamailio itself. It is quite self describing
On Tue, 7 Nov 2023, 18:18 SAMUEL MOYA TINOCO via sr-users, < sr-users@lists.kamailio.org> wrote:
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:
Asterisk
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
Soluciones inteligentes para la autonomía personal
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi Samuel,
First you need to verify at client , are you receiving 200k there, if 200 ok is received at client, then need to check why it unable to send ACK or sending on which ip, it seems Nating issue.
On Wed, Nov 8, 2023 at 3:30 AM SAMUEL MOYA TINOCO via sr-users < sr-users@lists.kamailio.org> wrote:
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:
Asterisk
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
Soluciones inteligentes para la autonomía personal
Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
-
Ihor Olkhovskyi -
James Browne -
M Arqum CH -
SAMUEL MOYA TINOCO -
Social Boh -
Yuriy G