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[gshergill at gltd.net]

Hi Amit,

My rtp.conf has the stunaddr listed and icesupport set to yes.

It looks like the issue is that the media isn't being sent from 192.168.3.150 to 192.168.3.131 (chrome browser to asteriskrtc.local).

When using asteriskrtc.local to originate the call (make a call directly from sipml client to another number on asteriskrtc.local or to a number on another asterisk server) audio flows both ways with no issue, it's just when asteriskgary.local is originating the call that there is no audio flowing from chrome to asteriskrtc.local.

I should probably rephrase the above though to say that on tshark I can actually see the packets flowing (tshark host 192.168.3.150):

2.384874 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
2.384925 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
2.385060 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175
2.385256 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021
2.394891 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021
2.415195 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021
2.434063 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
2.434121 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
2.434296 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175
2.434462 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021
2.435083 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021
2.455310 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021
2.475009 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021

Thanks again for your time!

Kind Regards,

Gary Shergill


----- Original Message -----
From: "Amit Patkar" <amit@avhan.com>
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 4:55:57 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)



Please check rtp.conf

Look for stunaddr setting. You can try with google STUN server
stunaddr = stun.l.google.com:19302





Thanks & Regards,
Amit Patkar
On 5/21/2014 9:13 PM, Gary Shergill wrote:


Hi again,

Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works...

Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting).

However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local).

Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way.

Thanks again.

Kind Regards,

Gary Shergill


----- Original Message -----
From: "Gary Shergill" <gshergill@gltd.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, May 21, 2014 3:36:54 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

Hi Amit,

ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix.

I have "directmedia=no" set for the user on asteriskrtc.local.

My exact setup/scenario is below:
- asteriskgary.local has a route to dial extensions on my Elastix server.
- asteriskgary.local has a route to dial extensions on asteriskrtc.local server.
- The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server.

What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000):
- User on asteriskgary.local places a call to 1000, his number is 6901
- 6901 answers on the web browser and begins to dial 1000
- 1000 answers and the call is established correctly
- SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...)
- 6901 can NEVER hear 1000

key:
192.168.3.127 - asteriskgary.local
192.168.3.131 - asteriskrtc.local
192.168.3.150 - machine running chrome browser where 6901 is logged on
192.168.3.100 - phone where 1000 is logged on

(1000 can hear 6901) RTP TRACE ON asteriskrtc.local
....
Got RTP packet from 192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160)
Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts 2304496624, len 000160)