Re: [asterisk-users] webrtc no audio

2015-08-28 Thread Vinicius Fontes 
I tested and it seems like I do have
https://issues.asterisk.org/jira/browse/ASTERISK-24146 but in a different
way. If I take more than 7s to answer the call, I don't get audio for a few
seconds (about 3), after that it works okay.



2015-08-28 10:43 GMT-03:00 Marek Červenka :

> are you sure you dont have this problem?
> https://issues.asterisk.org/jira/browse/ASTERISK-24146
>
> i'm now fighting with
> https://issues.asterisk.org/jira/browse/ASTERISK-24602
>
> Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):
>
> I have it working now!
>
> *I had to install Asterisk 13 with PJSIP support.That's important, even if
> you won't use PJSIP.* Then I did this configuration, which is working
> fine under NAT:
>
> *sip.conf:*
> [6001]
> type=friend
> secret=REDACTED
> host=dynamic
> context=interno
> disallow=all
> ;allow=alaw,h263,h264,vp8
> allow=g722
> dtmf=auto
> videosupport=yes
> transport=ws,udp
> avpf=yes
> callerid="WebRTC" <6001>
> encryption=yes
> qualify=yes
> directmedia=no
> nat=force_rport,comedia
> icesupport=yes
> dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
> dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
> dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
> DTLS cert file is
> dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
> DTLS private key is
> dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when
> setting up DTLS
>
> *rtp.conf:*
> icesupport=true
> stunaddr=stun.l.google.com:19302
>
> *res_stun_monitor.conf:*
> stunaddr = stun.l.google.com:19302; Address of the STUN server to
> query.
> stunrefresh = 30
>
> 2015-08-12 5:23 GMT-03:00 Marek Červenka :
>
>> 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?

>>>
>>> You have to do a packet capture, look at the exchange in Wireshark, and
>>> see how the negotiation flows. It requires a basic understanding of ICE.
>>>
>>>
>> it looks like we are facing this problem
>> 
>> https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
>> if we use "[]" in sipml5 expert config "To disable TURN/STUN to speedup
>> ICE candidates gathering you can use an empty array. e.g. []."
>> it works better
>>
>>
>>
>>
>> --
>> ---
>> Marek Cervenka
>> ===
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by 
>> http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
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Re: [asterisk-users] webrtc no audio

2015-08-28 Thread Marek Červenka 

are you sure you dont have this problem?
https://issues.asterisk.org/jira/browse/ASTERISK-24146

i'm now fighting with
https://issues.asterisk.org/jira/browse/ASTERISK-24602

Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):

I have it working now!

*I had to install Asterisk 13 with PJSIP support.That's important, 
even if you won't use PJSIP.* Then I did this configuration, which is 
working fine under NAT:


*sip.conf:*
[6001]
type=friend
secret=REDACTED
host=dynamic
context=interno
disallow=all
;allow=alaw,h263,h264,vp8
allow=g722
dtmf=auto
videosupport=yes
transport=ws,udp
avpf=yes
callerid="WebRTC" <6001>
encryption=yes
qualify=yes
directmedia=no
nat=force_rport,comedia
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when 
setting up DTLS


*rtp.conf:*
icesupport=true
stunaddr=stun.l.google.com:19302 

*res_stun_monitor.conf:*
stunaddr = stun.l.google.com:19302    
 ; Address of the STUN server to query.*

*
stunrefresh = 30

2015-08-12 5:23 GMT-03:00 Marek Červenka >:


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?


You have to do a packet capture, look at the exchange in
Wireshark, and see how the negotiation flows. It requires a
basic understanding of ICE.


it looks like we are facing this problem
https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
if we use "[]" in sipml5 expert config "To disable TURN/STUN to
speedup ICE candidates gathering you can use an empty array. e.g. []."
it works better




-- 
---

Marek Cervenka
===


-- 
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Marek Cervenka
===

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Re: [asterisk-users] webrtc no audio

2015-08-27 Thread Vinicius Fontes 
I have it working now!

*I had to install Asterisk 13 with PJSIP support.That's important, even if
you won't use PJSIP.* Then I did this configuration, which is working fine
under NAT:

*sip.conf:*
[6001]
type=friend
secret=REDACTED
host=dynamic
context=interno
disallow=all
;allow=alaw,h263,h264,vp8
allow=g722
dtmf=auto
videosupport=yes
transport=ws,udp
avpf=yes
callerid="WebRTC" <6001>
encryption=yes
qualify=yes
directmedia=no
nat=force_rport,comedia
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting
up DTLS

*rtp.conf:*
icesupport=true
stunaddr=stun.l.google.com:19302

*res_stun_monitor.conf:*
stunaddr = stun.l.google.com:19302; Address of the STUN server to query.
stunrefresh = 30

2015-08-12 5:23 GMT-03:00 Marek Červenka :

> 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?
>>>
>>
>> You have to do a packet capture, look at the exchange in Wireshark, and
>> see how the negotiation flows. It requires a basic understanding of ICE.
>>
>>
> it looks like we are facing this problem
> https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
> if we use "[]" in sipml5 expert config "To disable TURN/STUN to speedup
> ICE candidates gathering you can use an empty array. e.g. []."
> it works better
>
>
>
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
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Re: [asterisk-users] webrtc no audio

2015-08-12 Thread Marek Červenka 

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?


You have to do a packet capture, look at the exchange in Wireshark, 
and see how the negotiation flows. It requires a basic understanding 
of ICE.




it looks like we are facing this problem 
https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
if we use "[]" in sipml5 expert config "To disable TURN/STUN to speedup 
ICE candidates gathering you can use an empty array. e.g. []."

it works better




--
---
Marek Cervenka
===


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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Re: [asterisk-users] webrtc no audio

2015-08-11 Thread Joshua Colp 

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?


You have to do a packet capture, look at the exchange in Wireshark, and 
see how the negotiation flows. It requires a basic understanding of ICE.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
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Re: [asterisk-users] webrtc no audio

2015-08-10 Thread Vinicius Fontes 
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 
Date: 2015-07-27 13:54 GMT-03:00
Subject: No audio on SIP over WebRTC
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>


I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here's my relevant files.

*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes

[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass


*extensions.conf:*
[default]
exten => _6XXX,1,Dial(SIP/${EXTEN})


*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302



2015-08-10 12:35 GMT-03:00 Joshua Colp :

> Marek Cervenka wrote:
>
>> 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
>> (RTP flowing only in one direction)
>> BUT
>> call from person2(chrome) to person 3(Jitsi sip client) - works!
>>
>> any tips howto find the problem?
>>
>
> You would need to look at the ICE negotiation to see if it tried and
> failed. After that would be looking at the DTLS negotiation. Asterisk
> console output could provide some information.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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_
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Re: [asterisk-users] webrtc no audio

2015-08-10 Thread Joshua Colp 

Marek Cervenka wrote:

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
(RTP flowing only in one direction)
BUT
call from person2(chrome) to person 3(Jitsi sip client) - works!

any tips howto find the problem?


You would need to look at the ICE negotiation to see if it tried and 
failed. After that would be looking at the DTLS negotiation. Asterisk 
console output could provide some information.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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