Hello
I've succeeded in installing Asterisk 13 and more important : I can make
webRTC call and I have audio !!
For those on the search like myself, I want to spare some weeks of headache.
My steps (CentOS 6.8) :
yum install uuid-devel libuuid-devel autoconf patch automake
libcurl-devel
Hello
running into several problems when installing
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12).
I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled
First, I do not seem to have res_srtp module available, although all
necessary libs are
On 12-08-16 16:38, Joshua Colp wrote:
Jonas Kellens wrote:
Question : I noticed I received an error when installing pjproject
--with-external-srtp
I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")
Can this have anything to do with the
Jonas Kellens wrote:
Question : I noticed I received an error when installing pjproject
--with-external-srtp
I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")
Can this have anything to do with the no-audio-problems that I'm having ??
Question : I noticed I received an error when installing pjproject
--with-external-srtp
I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")
Can this have anything to do with the no-audio-problems that I'm having ??
Kind regards.
On
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-16 09:25, Антон Сацкий wrote:
Try delete nat from 77wrtc settings ice
Try delete nat from 77wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" wrote:
> On 11-08-16 18:03, Matt Fredrickson wrote:
>
>> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
>> wrote:
>>
>>> My main reason
On 11-08-16 18:03, Matt Fredrickson wrote:
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote:
My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache :-)
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
What in particular?
Any longer,
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
>
> I will do so if there is no other
I'm genuinely fascinated why you are insisting on using a version of
Asterisk almost 3 years old, for which EOL support ended last year.
Is there any particular reason you cannot or will not use the current
version as others have suggested?
Also, I see you are using Doubango and WebRTC, but in
Hello
Using Asterisk 12.8.2.
I now have the "via ICE" messages in the RTP debug (see below).
If you look in the SIP debug (see below), you also now see the
"ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the
webRTC client.
But still no audio ! None at all ! In both
My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This
can also cause headache :-)
I will do so if there is no other option.
But still, I don't see why Ast 13 would differ so much in this case ? If
ICE
My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse. WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.
Matthew Fredrickson
Hello
thank you for your answer.
I don't understand how there are many tutorials and examples on the web
where every time the outcome is a working setup. Very strange I feel now
after my personal experience with Asterisk 11 and webRTC.
You also say Asterisk 13. How about Asterisk 12 then ??
I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE. Are
you sure that the stun server you're using (the google one) still
works? I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut
On 10-08-16 08:52, Ludovic Gasc wrote:
For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
Hello
then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
This is no answer to my question.
So again : what am I
For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
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Hello
I'm trying for several days now to get ICE support for my Asterisk 11.23
on CentOS 6.
My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230
--> softphone Zoiper
(problem : no audio)
Reverse does not work either.
(problem : failed get local SDP)
I followed this
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