No audio in SIPML5 - Asterisk 11.13.1

[chyiannakou]

Hi,

I installed Asterisk 11.13.1 with Freepbx 2.11. I did the configuration on Asterisk for the WebRTC support following the https://code.google.com/p/sipml5/wiki/Asterisk quide and installed the Webrtc2sip gateway following the http://linux.autostatic.com/installing-webrtc2sip-on-ubuntu-1204. 

I do have outbound calls from SIPLM5 either in Chrome or Firefox but i have no audio when answering the call.

The webrtc2sip gateway runs when i execute ./webrtc2sip (see below)

I see there that the transport is ws://*10060 while on the Asterisk it is configured on port 8088. But if i change the config.xml file to 8088 and execute ./webrtc2sip i get errors. If i have the gateway started and perform the call i see not messages. So i am pretty sure that the media is not transferred through the gateway.  

Can somebody advice on this as i am not expert on this? 

root@debian:/usr/local/src/webrtc2sip# ./webrtc2sip

*******************************************************************

Copyright (C) 2012-2015 Doubango Telecom <http://www.doubango.org>

PRODUCT: webrtc2sip

HOME PAGE: http://webrtc2sip.org

LICENCE: GPLv3 or proprietary

VERSION: 2.6.0

'quit' to quit the application.

*******************************************************************

SSL is enabled :)

DTLS supported: yes

DTLS-SRTP supported: yes

*INFO: transport = udp://*:10060

*INFO: transport = ws://*:10060

*INFO: transport = wss://*:10062

*INFO: enable-rtp-symetric = yes

*INFO: enable-100rel = no

*INFO: enable-media-coder = no

*INFO: enable-videojb = yes

*INFO: video-size-pref = vga

*INFO: rtp-buffsize = 65535

*INFO: avpf-tail-length = [100-400]

*INFO: srtp-mode = optional

*INFO: srtp-type = sdes;dtls

*INFO: dtmf-type = rfc4733

*INFO: codecs = pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+

*INFO: UnRegister codec: PCMA, G.711a codec (native)

*INFO: UnRegister codec: PCMU, G.711u codec (native)

*INFO: UnRegister codec: GSM, GSM Full Rate (libgsm)

*INFO: UnRegister codec: VP8, VP8 codec (libvpx)

*INFO: UnRegister codec: H264, H264 Base Profile (FFmpeg, x264)

*INFO: UnRegister codec: H264, H264 Main Profile (FFmpeg, x264)

*INFO: UnRegister codec: H263, H263-1996 codec (FFmpeg)

*INFO: UnRegister codec: H263-1998, H263-1998 codec (FFmpeg)

*INFO: codec-opus-maxrates = 48000;48000

*INFO: stun-server = stun.l.google.com;19302;-;-

*INFO: enable-icestun = yes

*INFO: max-fds = -1

*INFO: transport = c2c://*:10070

*INFO: transport = c2cs://*:10072

*INFO: database = sqlite;*

*INFO: sqlite3_threadsafe = 1

*INFO: Database opened = TRUE

*INFO: tnet_transport_prepare()

*INFO: pipeR fd=16, pipeW=17

*INFO: Socket added[TCP/IPv4 transport]: fd=16, tail.count=1

*INFO: master fd=10

*INFO: Socket added[TCP/IPv4 transport]: fd=10, tail.count=2

*INFO: Transport::run(TCP/IPv4 transport) - enter

*INFO: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {10070} using fd {10} with type {9}...

*INFO: tnet_transport_prepare()

*INFO: pipeR fd=18, pipeW=19

*INFO: Socket added[TLS/IPv4 transport]: fd=18, tail.count=1

*INFO: master fd=11

*INFO: Socket added[TLS/IPv4 transport]: fd=11, tail.count=2

*INFO: Transport::run(TLS/IPv4 transport) - enter

*INFO: Stack running in SERVER mode

*INFO: tsk_timer_manager_start

*INFO: Starting [TLS/IPv4 transport] server with IP {0.0.0.0} on port {10072} using fd {11} with type {17}...

*INFO: Timer manager run()::enter

*INFO: TIMER MANAGER -- START

*INFO: SIP STACK::run -- START

*INFO: tnet_transport_prepare()

*INFO: pipeR fd=23, pipeW=24

*INFO: Socket added[SIP transport]: fd=23, tail.count=1

*INFO: master fd=20

*INFO: Socket added[SIP transport]: fd=20, tail.count=2

*INFO: tnet_transport_prepare()

*INFO: pipeR fd=25, pipeW=26

*INFO: Socket added[SIP transport]: fd=25, tail.count=1

*INFO: master fd=21

*INFO: Socket added[SIP transport]: fd=21, tail.count=2

*INFO: tnet_transport_prepare()

*INFO: pipeR fd=27, pipeW=28

*INFO: Socket added[SIP transport]: fd=27, tail.count=1

*INFO: master fd=22

*INFO: Socket added[SIP transport]: fd=22, tail.count=2

*INFO: Transport::run(SIP transport) - enter

*INFO: SIP STACK -- START

*INFO: Transport::run(SIP transport) - enter

*INFO: Starting [SIP transport] server with IP {192.168.0.130} on port {10060} using fd {21} with type {64}...

*INFO: Transport::run(SIP transport) - enter

*INFO: Starting [SIP transport] server with IP {192.168.0.130} on port {10060} using fd {20} with type {2}...

*INFO: Starting [SIP transport] server with IP {192.168.0.130} on port {10062} using fd {22} with type {128}...

[navaismo]

You need to choose if your asterisk will work with or without webRTC settings, if you are using the media gateway then in your asterisk you configure the extensions as usually you do, if you configure your asterisk with webrtc settings(DTLS_SRTP) then you dont need the media gateway. 

[chyiannakou]

Thank you for your answer.

Initially i configured my Asterisk with DTLS_SRTP but i did not have audio when calling.

After reading some posts through this group, i understood that the media gateway is also a must. That's why i proceed to install it.

So in my case i suspend that i use the DTLS_SRTP configuration.

Any ideas why i have no audio in the case of DTLS_SRTP configuration? Where shall i look in order to further investigate why i have no audio streams when calling out.

Thank you again

Chris

[navaismo]

You need to look at the SDP transaction, SIP, RTP and JS debugs.

[Mamadou DIOP]

if your Asterisk installation supports WebRTC features then, you don't need webrtc2sip. In such case if you have audio issues you should ask on Asterisk forum.

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[chyiannakou]

Hi again,

If i choose to use the webrtc2sip gateway what are the parameters that i have to put in the SIPML5 configuration? I am really confused.

Thank you

On Tuesday, February 24, 2015 at 5:31:22 PM UTC+2, navaismo wrote:

[chyiannakou]

Hi again,

In the JS debug i see the following errors:

Not implementedsrc/tinySAK/src/tsk_utils.js?svn=222:128 tsk_utils_log_error

src/tinySAK/src/tsk_utils.js?svn=222:128 Failed to find transportsrc/tinySAK/src/tsk_utils.js?svn=222:128 tsk_utils_log_error

src/tinySAK/src/tsk_utils.js?svn=222:128 Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.src/tinySAK/src/tsk_utils.js?svn=222:128 tsk_utils_log_error

tsk_utils.js?svn=222:128 Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.tsk_utils.js?svn=222:128 tsk_utils_log_errortmedia_session_jsep.js?svn=222:691 tmedia_session_jsep01.onSetRemoteDescriptionErrortmedia_session_jsep.js?svn=222:870 (anonymous function)

3tsk_utils.js?svn=222:128 Not implementedtsk_utils.js?svn=222:128 tsk_utils_log_errortsip_dialog_layer.js?svn=222:202 tsip_dialog_layer.handle_incoming_messagetsip_transport_layer.js?svn=222:231 tsip_transport_layer.handle_incoming_messagetsip_transport.js?svn=222:425 __tsip_transport_ws_onmessage

Can anybody guide me where to look in order to correct these issues?

Thank you

Chris

[navaismo]

This:

Called with SDP without ice-ufrag and ice-pwd

usually means thta your asterisk wasn´t compiled with webrtc support. You need to add all dependencies and recompile again.

[chyiannakou]

Hi Navaismo,

I followed the wiki https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 manual in order to install all the dependencies and recompiled again.

But again i get the same error on the JS debug. So the issue must be somewhere else...

I followed your article "Troubleshoot WebRTC issues"

The only issue i found according to your guide is that in the RTP debug i don't see (VIA ICE) in the sent SRTP packages. See below
Got  RTP packet from    192.168.0.126:10020 (type 00, seq 007202, ts 000160, len 000160)
Sent RTP packet to      192.168.0.126:60736 (type 00, seq 012928, ts 000160, len 000160)

According to this you mention that Asterisk does not have the ICE support enabled in the peer or in the rtp.conf.

I do have the option icesupport =true in the rtp_custom.conf ( i use freepbx)
And also have the option icesupport =yes in the webrtc extension 4000.

Can you advice what i miss here?

Thank you

Chris

[navaismo]

Or your build itself doesnt have ICE support. The lack of ufrag is because asterisk can handle it.

[chyiannakou]

Hi again,

I updated my FreePBX with the latest Asterisk 11 current (version 11.16.0) and everything works ok now.

Thank you both Navaismo and Mamadou for all your help

Christodoulos

[chyiannakou]

Hi again Navaismo,

Suddenly i face the no audio issue (no audio in either site) with my Debian based FreePBX.
Follwing your troubleshoot guide http://forums.digium.com/viewtopic.php?p=199275 everything seems to be ok.
I use Firefox 37.0.2 and Chrome 42.0.2311.90.

any advice where to look for this?

Christodoulos

[navaismo]

Update to asterisk 11.16 and try again.

[chyiannakou]

Hi,

But i am already in Asterisk 11.16. I upgraded to Asterisk 11.16 on March the 4th and was working then.. Now it does not without performing any changes.....

[chyiannakou]

Hi again,

The WebRTC seems to work only with Google Chrome not with Firefox. Is this correct?

And one more question.

I have two identical FreePBX systems, one in a local machine and one in a VPS server accessible by VPN connection.
When i test the WebRTC functionality from Chrome over the local machine it works ok.
When i test the WebRTC functionality from Chrome over the VPS server, the Audio is sent without ICE support (i checked the RTP packages over the CLI), therefore i have no audio in this case although the call is connected.
As already mentioned the two systems are identical, same OS, same Asterisk and FreePBX versions, same configuration.

Is there any possibility the ICE support to be blocked by the Private network? What shall i check for this?

Thank you

Chris

[Gopalakrishnan N]

Initially I too faced the same issue. But after installing latest Asterisk 11.x current, I gues 11.16.0, it worked fine. No am planning to upgrade all my production servers to this version. 

Please try with latest current Asterisk and latest Sipml5. 

Regards,

Gopal. 

[chyiannakou]

I am already in the latest Asterisk 11 current (11.17.1) and the latest SIPML5 . Do i miss any package?

[chyiannakou]

Hi Again,

The error message i get is the following:
PJ ICE Rx error status code: 370401 'Bad Request'.

From the SIP debug file (it is attached0 i see that ICE support is enabled. See line 202

a=ice-ufrag:ex/b13jTvuKEX5kE
a=ice-pwd:BiKsem3CSxzZ55xBI42DUaMG

Can anybody help with this error?

Christodoulos

[Gopalakrishnan N]

Try disabling RTCwebbreaker in the expert mode. 

Thanks.