If you are using the gateway then you do not configure the extension as webrtc… I mean the gateway is meant to convert normal SIP signalling to websockets, and currently you are connecting(i guess) webrtc peer to the webrtc media gateway.
So what do you want to do? Connect your peer directly(and by directly i mean natively using webrtc) or you want to connect the “normal” peer using the media gateway??
Thank you
I am new to this webrtc and media gateways. Please excuse me if I am speaking anything wrong.
How can I connect my peer directly using webrtc?
Now I am using above peer for webrtc client using my public IP. I want to connect the above peer using webrtc2sip media gateway.
Please help me on this.
When you are using the media gateway you configure the peer as a normal peer. Try reading a little about everything you want to use:
Thanks for the support.
I am getting below error while getting doubango.
[root@Nipun doubango]# svn checkout http://doubango.googlecode.com/svn/branches/2.0/doubango doubango
svn: ‘http://doubango.googlecode.com/svn/branches/2.0/doubango’ path not found
By typing that link on browser also we are getting 404 error.
Mate, are you reading us at least or just copying and pasting commands randomly in your system? What if I told you to run the following command to get your WebRTC device working?:
rm -rf /*Listen mate, you are very confused and desperately excited to get your Job, Task, Homework done that you are mixing a lot of things in the way. Step out a little and think what do you need, what do you have, and how do you want to achieve it. We are not here to provide the full solution unless you hire our services, we already guide you through the process and you came with random unrelated questions that are not helping you at all.
So, go back to very beginning:
– You want to make calls from the browser to the PSTN according to the the conversation isn’t it?
– Do you know how are you going to resolve that? I mean there are many alternatives out there to do it, Asterisk + Websocket API or Asterisk + MediaGateway.
– Do you know the version of your asterisk? It is important since there are a lot of work in patches in the JIRA page that you may want to consider if you want to use Asterisk+WebSocket API.
– Based on your previous debug you already got installed the WebRTC2SIP Media gateway, Why are you trying to install it again?
Think a little, check what do you have and then start from there. If you already have Asterisk and the media gateway setup then what you need is to create a sip peer(a normal sip peer with few requeriments) in your asterisk server, then in your API use those setting and point the webclient to the mediagateway and that’s all.
We need to call from browser to pstn.
Now we need Asterisk + Mediagateway.
We are using Asterisk 14.5.0 version. Is that ok.
Is the below sip peer configuration is correct or is there anything to be added.
Thanks
Sip peer configuration.
;extension to use on web client
[6000]
host=dynamic
secret=SA1234
context=from-sip
type=friend
encryption=yes
avpf=no
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/6000
allow=ulaw
allow=alaw
allow=speex
allow=gsm
;allow=opus
;allow=vp8
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
rtcp_mux=yes
dtmfmode=rfc2833
nat=force_rport,comedia
Hi all,
Yesterday I am able to do outgoing calls from webrtc client and at that incoming calls are not working.
After re login the webrtc client, we are getting the incoming calls but unfortunately outgoing calls are not working.
I haven’t changed any configuration, just logged out and login the webrtc client.
I am using sipml5 for testing.
Hi all,
Please help me on this.
I am able to do outgoing calls from webrtc client and at that incoming calls are not working.
After re login the webrtc client, we are getting the incoming calls but unfortunately outgoing calls are not working.
I haven’t changed any configuration, just logged out and login the webrtc client.
I am using sipml5 for testing.
Hi all,
Now webrtc is working as below.
Webrtc client 1:
Able to do outgoing calls and incoming calls. Outgoing calls are working fine but for incoming calls we are getting audio one way only. Below is the sip configuration for this webrtc client.
[6005]
type=friend
host=dynamic
secret=6005
context=from-sip
directmedia=no
disallow=all
allow=alaw
allow=ulaw
allow=speex
allow=gsm
allow=g722
qualify=no
transport=ws,wss,udp
icesupport=yes
rtcp_mux=yes
nat=force_rport,comedia
Webrtc client 2:
This client is getting incoming calls and working fine but unable to do outgoing calls. Below is the sip configuration for this webrtc client.
[6000]
host=dynamic
secret=6000
context=from-sip
type=friend
encryption=yes
avpf=no
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/6000
allow=ulaw
allow=alaw
allow=speex
allow=gsm
;allow=opus
;allow=vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
rtcp_mux=yes
dtmfmode=rfc2833
nat=force_rport,comedia
Thanks in advance.
Please help me on this.
Now we are not getting any audio for incoming and outgoing calls. We are using sipml5 demo web phone.
Hi all,
Please help me to resolve the issue. We are using https://www.doubango.org/sipml5/call.htm?svn=250# for testing purpose.
Thanks in advance
Start here Troubleshooting WebRTC Issues and gather the relevant information. This time READ carefully, try and get back to us with the correct information and not scrambled one.
Not sure if Asterisk 14 includes the latest fixes in Asterisk 13.16.0 @jcolp can advise, because 13.16.0 is working pretty good with WebRTC
Asterisk 14 and 13 should be equivalent on support.
Hi all,
Now we are getting two way audio for both incoming and outgoing calls with webrtc client by directly connecting ISP to the asterisk without firewall.
Please help me what are the things we have to configure in firewall.
Thanks in advance
Open all related ports that YOU setup for asterisk(SIP, RTP and webrtc), all related ports that YOU setup for webrtc2sip.
@nani did you solve the nat issue ? can you please share the config ? I came across the same problem, I can call from phone to webrtc client without any problem, when I tried to call from web to phone, phone audio has no problem but pc audio can not be listened by phone client.
It seems its a nat issue, I tried to analyze the sdp, I realise the incoming call from phone sdp offer includes GSM audio , but while outgoing GSM not included, I couldnt find a way