Displaying 20 results from an estimated 2000 matches similar to: "Cannot get my first WebRTC experiment to work."
2015 Jan 26
0
Need help interpreting SDP on failing WebRTC connection
Hi,
I am trying to setup a WebRTC connection to asterisk 1.13.0.
Using Bria a regular SIP connection works, but using sipml5 on chrome, I
got nothing.
My network setup by the way: I am working behind a comcast cable modem, the
test setup is at digital ocean, and from my laptop I also have a direct VPN
connection
to the asterisk server my laptop being 192.168.241.10 and asterisk being
2015 Jan 27
0
Need some help interpreting SDP on a failing WebRTC connection
Hi,
I am trying to setup a WebRTC connection to asterisk 1.13.0.
Using Bria a regular SIP connection works, but using sipml5 on chrome, I
got nothing.
My network setup by the way: I am working behind a comcast cable modem, the
test setup is at digital ocean, and from my laptop I also have a direct VPN
connection
to the asterisk server my laptop being 192.168.241.10 and asterisk being
2014 May 10
2
Asterisk 11.9 with webRTC demo integration
Hi All,
I am trying to configure webRTC phone example for SIPml5 and i found this
info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
.
I have asterisk 11.9.0 installed and downloaded source of SIPml5 from
http://code.google.com/p/sipml5/source/checkout I copied sample code into
web root directory and example loaded successfully and also able to
register 2 extensions.
I
2016 Feb 18
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie
station.
Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode
form (see [1]), I'm getting this error :
*2:SecurityError: Failed to construct
2014 Jul 02
1
Webrtc Not acceptable here
Hi,
I am getting
*Can't provide secure audio requested in SDP offer*
with sipml5 client hosted on my local system
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=sameer ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF
2016 Feb 18
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>:
> my experience with pjsip for webrtc
> http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
>
>
> Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use Live demo (
2015 Jun 16
1
Req help regarding webRTC : Attempted Attempted to set an invalid DTLS-SRTP configuration on RTP instance
Hi List,
I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I
have installed Asterisk 13.4 with srtp,pjproject. I have configured two
numbers for webRTC clients, when i try to call from a client (sipml5) to
another client (sipml5) it throws the following error:
"chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid
DTLS-SRTP configuration on RTP
2019 Jan 04
2
CyberMegaPhone WebRTC Video Conference demo
I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017
I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K.
When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo).
After
2014 Apr 14
1
Webrtc and adventures with Asterisk 11
Hi,
I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 +
opus/vb8 codec patch. This is interesting technology and I try to find
out how to connect all the moving parts.
Firefox:
Neither sipml5 or jssip works with calls to asterisk, audio/video
doesn't matter.
WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream
without encryption details: audio 35684
2015 Mar 12
2
WebRTC demo phones
Hello,
Can anyone recommend a particular online WebRTC phone for testing with
Asterisk?
We tried:
- JsSIP, but even with the "enable video" checkbox disabled it sends video
options in the INVITE SDP and Asterisk rejects it with "Rejecting secure
video stream without encryption details".
- sipML5, but it won't register, perhaps something to do with not using the
Asterisk
2014 Nov 12
1
Como unir webrtc con asterisk???
tengo la siguiente pagina pero no se como seguir despues del punto 22
http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html
gracias!
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2015 Aug 10
2
webrtc no audio
hello,
i'm facing strange problem
asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
person1 to person3 are behind different NATs
audio devices double checked
call from person1(chrome) to person2(chrome) works
call from person1(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side
2015 Aug 11
2
webrtc no audio
I'm having the same issue! The difference in my case is Asterisk server has
a public IPv4 and the browser is behind a single NAT.
I'm forwarding my configuration below (which I posted previously on
asterisk-users).
How can we debug ICE negotiation?
---------- Forwarded message ----------
From: Vinicius Fontes <vinicius at aittelecom.com.br>
Date: 2015-07-27 13:54 GMT-03:00
2015 Aug 12
2
webrtc no audio
Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
> Vinicius Fontes wrote:
>> I'm having the same issue! The difference in my case is Asterisk server
>> has a public IPv4 and the browser is behind a single NAT.
>>
>> I'm forwarding my configuration below (which I posted previously on
>> asterisk-users).
>>
>> How can we debug ICE negotiation?
>
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
2016 Sep 08
3
Asterisk 13 and WebRTC
Hello list,
before to lost my time, I'd like know if someone have a WebRTC working
configuration on Asterisk 13.11.0 SIP or PJSIP channel.
Thank you
Regards
2012 Aug 07
1
Asterisk & Websockets
Hi everyone,
I'm currently trying to play a little with WebRTC using sipml5 client and
Asterisk trunk (370821)
It seems the the WebRTC implementation for Asterisk 11 is already available
in the trunk? Am I right?
http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html
I'm having trouble to even register to my Asterisk server using sipml5
client.
The only reference to
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
2014 Mar 22
0
webrtc not working with asterisk 11.8 + jssip/sipml5
users are registering over ws:// but while dialing A -> B , using either
jssip/sipml5
I receive an error on B side saying , ice related information is missing ,
and in INVITE sdp mentioned fields are really missing. Exact error in
console is
SetRemoteDescription failed: Called with an SDP without ice-ufrag and
ice-pwd
lots of users are using webrtc with jssip/sipml5 + asterisk successfully.
2012 Dec 17
1
[webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Dear All,
I use sipml5 to register two users from browser and the two clients are
successfully connected. But when I made a call from one of the users, the
other user doen'st have call notification and for a while the calling
process ended. I check the /var/log/asterisk/messages got the following log:
[Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF
profle in audio