15 Jun
2018
15 Jun
'18
12:48 a.m.
Hi everybody, I followed this tutorial
https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamail…
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.
15 Jun
15 Jun
1:25 a.m.
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 :)
1:37 a.m.
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(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
1:40 a.m.
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(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:29 p.m.
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(a)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(a)pocos.nl> wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel
Gianico wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
5:30 p.m.
Or maybe FreeSwitch is redundant if you use rtpengine…
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Emanuel Gianico
Sent: fredag 15. juni 2018 13:29
To: Kamailio (SER) - Users Mailing List <sr-users(a)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<mailto: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<mailto: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<mailto: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<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
21 Jun
21 Jun
8:35 p.m.
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(a)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(a)lists.kamailio.org> * On Behalf Of *Emanuel
Gianico
*Sent:* fredag 15. juni 2018 13:29
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)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(a)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(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
9:59 p.m.
https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/webso…
2018-06-21 18:35 GMT+03:00 Emanuel Gianico <emanuelgianico(a)gmail.com>om>:
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(a)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(a)lists.kamailio.org> * On Behalf Of *Emanuel
Gianico
*Sent:* fredag 15. juni 2018 13:29
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)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(a)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(a)pocos.nl> wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
15 Jun
15 Jun
1:21 p.m.
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
3:18 p.m.
On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote:
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.
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.
5:26 p.m.
On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan
Christensen wrote:
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
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
2137
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10 comments
5 participants
participants (5)
-
Daniel Tryba -
David Villasmil -
Emanuel Gianico -
Pan Christensen -
Yuriy Gorlichenko