Displaying 20 results from an estimated 6000 matches similar to: "Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work"
2014 Dec 05
0
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
On 05/12/14 16:46, Olli Heiskanen wrote:
> INVITE that Asterisk (at port 5070) receives:
> PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046
> INVITE sip:660 at testers.com
> <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0
> Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177>
> Via: SIP/2.0/UDP
>
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this
2014 Aug 11
1
Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk changes them in the sdp bodies along the way. I'm using Asterisk
11.11.0.
Is there a way to
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
Hello,
I was testing with sdp and something came up worth asking:
While calling from a webrtc client to another (chrome, sip.js) Asterisk
receives the following sdp and rejects it with 488 Not Acceptable. Why does
this happen, what's wrong with the sdp? The second sdp body below is
accepted instead. Both have rtp profile RTP/SAVPF, difference is that the
second one was produced by rtpengine,
2015 Feb 26
2
WebRTC phone
Can anyone recommend a good WebRTC phone to use with Asterisk? I do
not mind if it is commercial or open source. Customers are starting to
ask for web solutions and we need to start testing.
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez
+52 (55)9116-91161
2015 Feb 26
0
WebRTC phone
For the client:
JSSIP and Sipml5.
If you are going to be coding something up yourself I like the JSSIP 0.5.x
javascript interfaces. If you are simply going to use a pre-canned one then
sipml5 works pretty well and remembers your settings in localstorage. I
haven't used any closed source versions since the above works really well
for us.
For the server:
If you are using Asterisk 1.8
2015 Mar 04
0
WebRTC phone
On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote:
> For those that were interested I have attached the kamailio.cfg which we
> have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
> following yum packages:
>
> kamailio.x86_64 4.2.1-4.1
> @home_kamailio_v4.2.x-rpms
> kamailio-auth-ephemeral.x86_64
2015 Mar 04
2
WebRTC phone
For those that were interested I have attached the kamailio.cfg which we
have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
following yum packages:
kamailio.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-auth-ephemeral.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-bdb.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
2006 Mar 06
1
Announce of our Xen project; Routing simulation with Einar
Hello all,
Some time before Christmas, a project was started. The task was to
enhance an existing LiveCD solution with User-mode Linux. A number of
requirements was stated, where the greatest challenge was to fit 10
virtual routers within an old no-name PC with only 256MB Ram. And the
solution must still be run from a LiveCD.
Thanks to Xen, the task was possible to complete.
-But, what does it
2013 Jun 17
1
Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017)
and get a SIP 488 Not Acceptable Here response.
I have no problems using the same Asterisk configuration and the same page
to make a call from Chrome.
I have seen other people post a similar issue, but I have not seen a
solution. If someone with good knowledge of this issue were to respond
with "this is a known
2014 Aug 06
1
From and To headers contain same account in INVITEs
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I have 660 at testers.com as
a websocket client and 700 at testers.com (caller) using a Zoiper client (db
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
Hi All.
I'm running some tests with the latest Asterisk SVN-branch-12-r410493M
compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS
machine (2.6.32-358.18.1.el6.i686).
As a client I'm using the sipMLP WebRTC javascript softphone running on
Chrome 33.0.1750.146 m.
I have the softphone correctly registered on the Asterisk machine but as
soon as I try to start a new call
2014 Aug 25
0
WebRTC / Rejecting secure audio stream errors
I've seen the following appear in some tests with Asterisk 11.11:
WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting
secure audio stream without encryption details: audio 54908
UDP/TLS/RTP/SAVPF 109 0 8 101
Specifically, it always happens from a Firefox 24 host but it works
without this error from another host running Firefox 26
I did a diff on the SDP and couldn't see
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root at elx4 ~]#
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2015 May 21
1
asterisk 13 webrtc
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in
SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2004 Oct 05
2
Problems installing app_valetparking
I download app_valetparking.c from
http://www.loligo.com/asterisk/misc/apps/app_valetparking.c
and followed the directions on
http://www.loligo.com/asterisk/misc/apps/app_valetparking.README
I am using asterisk-1.0.0 any suggestions
[root@localhost asterisk]# astxs -install apps/app_valetparking.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat
ions -g -Iinclude
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video