Hi everybody, I followed this tutorial
https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamaili...
And it works fantastic!
The next step was to add WebRTC support, so I added WebSockets module to enable web clients to register on kamailio. It works flawlessly and webphone clients register OK! (Followed this http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket )
Now, when I call from a softphone (eyeBeam) to the web client (jssip) the other party reach okay, rings okay, but when I pickup de call (from the web client), the softphone goes directly to the VoiceMail.
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
I would like to avoid RTPEngine, because from what I understand, FreeSwitch can handle the media part.
Can somebody please have the kindness to guide me on how to enable webrtc between Kamailio and FreeSwitch? If somebody needs to see the "kamailio.cfg" please let me know, and i would upload the file to a gist.
Cheers, Emanuel.
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
I would like to avoid RTPEngine, because from what I understand, FreeSwitch can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Yeah, you need to set the correct offer, i did that a while ago, but i can't remember how i did it.
Check out https://github.com/havfo/WEBRTC-to-SIP
Hope it help.
David
On Thu, Jun 14, 2018, 22:26 Daniel Tryba d.tryba@pocos.nl wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."
I would like to avoid RTPEngine, because from what I understand,
FreeSwitch
can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba d.tryba@pocos.nl wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."
I would like to avoid RTPEngine, because from what I understand,
FreeSwitch
can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I'm going to investigate Kazoo samples as Gorlichenko suggested because I think using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch fully supports WebRTC
El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko ovoshlook@gmail.com escribió:
You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba d.tryba@pocos.nl wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."
I would like to avoid RTPEngine, because from what I understand,
FreeSwitch
can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Or maybe FreeSwitch is redundant if you use rtpengine…
With kind regards Pan B. Christensen Developer Phonect AS
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Emanuel Gianico Sent: fredag 15. juni 2018 13:29 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Kamailio + FreeSwitch + WebRTC
I'm going to investigate Kazoo samples as Gorlichenko suggested because I think using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch fully supports WebRTC
El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko <ovoshlook@gmail.commailto:ovoshlook@gmail.com> escribió: You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> wrote: On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
I would like to avoid RTPEngine, because from what I understand, FreeSwitch can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I don't think so... The idea is to use FS as a media server. Why I would use RTPEngine? FS offers the same and more:
- WebRTC support - VoiceMail Server - Queues - IVR - Announcements
And so on...
The idea is to use Kamailio in front and throw all media related stuff to FS.
There is a way to accomplish this? I searched a lot but I couldn't find nothing about it, only about Kamailio and RTPProxy (or RTPEngine)
Best regards, Emanuel.
El vie., 15 de jun. de 2018 09:32, Pan Christensen < pan.christensen@phonect.no> escribió:
Or maybe FreeSwitch is redundant if you use rtpengine…
With kind regards *Pan B. Christensen* Developer Phonect AS
*From:* sr-users sr-users-bounces@lists.kamailio.org * On Behalf Of *Emanuel Gianico *Sent:* fredag 15. juni 2018 13:29 *To:* Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject:* Re: [SR-Users] Kamailio + FreeSwitch + WebRTC
I'm going to investigate Kazoo samples as Gorlichenko suggested because I think using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch fully supports WebRTC
El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko ovoshlook@gmail.com escribió:
You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba d.tryba@pocos.nl wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."
I would like to avoid RTPEngine, because from what I understand,
FreeSwitch
can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/ kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websoc...
2018-06-21 18:35 GMT+03:00 Emanuel Gianico emanuelgianico@gmail.com:
I don't think so... The idea is to use FS as a media server. Why I would use RTPEngine? FS offers the same and more:
- WebRTC support
- VoiceMail Server
- Queues
- IVR
- Announcements
And so on...
The idea is to use Kamailio in front and throw all media related stuff to FS.
There is a way to accomplish this? I searched a lot but I couldn't find nothing about it, only about Kamailio and RTPProxy (or RTPEngine)
Best regards, Emanuel.
El vie., 15 de jun. de 2018 09:32, Pan Christensen < pan.christensen@phonect.no> escribió:
Or maybe FreeSwitch is redundant if you use rtpengine…
With kind regards *Pan B. Christensen* Developer Phonect AS
*From:* sr-users sr-users-bounces@lists.kamailio.org * On Behalf Of *Emanuel Gianico *Sent:* fredag 15. juni 2018 13:29 *To:* Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject:* Re: [SR-Users] Kamailio + FreeSwitch + WebRTC
I'm going to investigate Kazoo samples as Gorlichenko suggested because I think using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch fully supports WebRTC
El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko ovoshlook@gmail.com escribió:
You can watch at the kazoo project examples if you want to avoid rtp proxy
On Thu, Jun 14, 2018, 23:26 Daniel Tryba d.tryba@pocos.nl wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."
I would like to avoid RTPEngine, because from what I understand,
FreeSwitch
can handle the media part.
IIRC I got the same error in my tries to transcode/bridge SIP over TLS with SRTP to just plain old SIP with RTP. I haven't put any effort in it to get it working. You'll need to play around with rtpengine offer/answer, I based my test on https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamai lio/kamailio.cfg but I blamed my failure on an old rtpengine :)
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I based my test on https://github.com/havfo/WEBRTC-to- SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
You need to add 'SDES-off' to the rtpengine_manage strings for calls going to WebRTC. Most browsers don't support fallback to SDES (anymore) and will reject the call if both DTLS and SDES are offered.
With kind regards Pan B. Christensen Developer Phonect AS
On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote:
I based my test on https://github.com/havfo/WEBRTC-to- SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
You need to add 'SDES-off' to the rtpengine_manage strings for calls going to WebRTC. Most browsers don't support fallback to SDES (anymore) and will reject the call if both DTLS and SDES are offered.
IIRC the problem I had was that going from RTP to SRTP, there was no key exchange in SDP (a=crypto) being added by rtpengine. I'll look into this again in the near future.
On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote:
I based my test on https://github.com/havfo/WEBRTC-to- SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
You need to add 'SDES-off' to the rtpengine_manage strings for calls going
to WebRTC. Most browsers don't support fallback to SDES (anymore) and will reject the call if both DTLS and SDES are offered.
IIRC the problem I had was that going from RTP to SRTP, there was no key exchange in SDP (a=crypto) being added by rtpengine. I'll look into this again in the near future.
a=crypto is for SDES, and should not be present in SDP. You need a=fingerprint for DTLS. Currently both are added by rtpengine if RTP/SAVPF is specified without SDES-off
With kind regards Pan B. Christensen Developer Phonect AS
-
Daniel Tryba -
David Villasmil -
Emanuel Gianico -
Pan Christensen -
Yuriy Gorlichenko