[asterisk-users] One Way Audio with WebRTC (with external asterisk)

Wed May 21 09:36:54 CDT 2014

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)
       > 0x7fe73c021740 -- Probation passed - setting RTP source address to 192.168.3.127:15942
Got  RTP packet from    192.168.3.127:15942 (type 00, seq 003763, ts 000160, len 000160)
Sent RTP packet to      192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 000160, len 4294967284)
Got  RTP packet from    192.168.3.150:55148 (type 00, seq 014309, ts 2304496791, len 000160)
Sent RTP packet to      192.168.3.127:15942 (type 00, seq 054710, ts 2304496784, len 000160)
Got  RTP packet from    192.168.3.150:55148 (type 00, seq 014310, ts 2304496951, len 000160)
Sent RTP packet to      192.168.3.127:15942 (type 00, seq 054711, ts 2304496944, len 000160)
Got  RTP packet from    192.168.3.127:15942 (type 00, seq 003764, ts 000320, len 000160)
Sent RTP packet to      192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 000320, len 4294967284)
Got  RTP packet from    192.168.3.150:55148 (type 00, seq 014311, ts 2304497111, len 000160)
Sent RTP packet to      192.168.3.127:15942 (type 00, seq 054712, ts 2304497104, len 000160)
Got  RTP packet from    192.168.3.127:15942 (type 00, seq 003765, ts 000480, len 000160)
Sent RTP packet to      192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 000480, len 4294967284)
....

(1000 can hear 6901) RTP TRACE ON asteriskgary.local
...
Got  RTP packet from    192.168.3.131:17836 (type 00, seq 055375, ts 2304603184, len 000160)
Sent RTP packet to      192.168.3.131:17836 (type 00, seq 004428, ts 106560, len 000160)
Got  RTP packet from    192.168.3.131:17836 (type 00, seq 055376, ts 2304603344, len 000160)
Sent RTP packet to      192.168.3.131:17836 (type 00, seq 004429, ts 106720, len 000160)
Got  RTP packet from    192.168.3.131:17836 (type 00, seq 055377, ts 2304603504, len 000160)
Sent RTP packet to      192.168.3.131:17836 (type 00, seq 004430, ts 106880, len 000160)
Got  RTP packet from    192.168.3.131:17836 (type 00, seq 055378, ts 2304603664, len 000160)
Sent RTP packet to      192.168.3.131:17836 (type 00, seq 004431, ts 107040, len 000160)
...

(no audio) RTP TRACE ON asteriskrtc.local
....
Got  RTP packet from    192.168.3.127:17796 (type 00, seq 035016, ts 000640, len 000160)
Sent RTP packet to      192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 000640, len 4294967284)
Got  RTP packet from    192.168.3.127:17796 (type 00, seq 035017, ts 000800, len 000160)
Sent RTP packet to      192.168.3.150:53684 (via ICE) (type 00, seq 060982, ts 000800, len 4294967284)
Got  RTP packet from    192.168.3.127:17796 (type 00, seq 035018, ts 000960, len 000160)
Sent RTP packet to      192.168.3.150:53684 (via ICE) (type 00, seq 060983, ts 000960, len 4294967284)
Got  RTP packet from    192.168.3.127:17796 (type 00, seq 035019, ts 001120, len 000160)
Sent RTP packet to      192.168.3.150:53684 (via ICE) (type 00, seq 060984, ts 001120, len 4294967284)
Got  RTP packet from    192.168.3.127:17796 (type 00, seq 035020, ts 001280, len 000160)
Sent RTP packet to      192.168.3.150:53684 (via ICE) (type 00, seq 060985, ts 001280, len 4294967284)
....

(no audio) RTP TRACE ON asteriskgary.local
...
Sent RTP packet to      192.168.3.131:16116 (type 00, seq 035355, ts 054880, len 000160)
Sent RTP packet to      192.168.3.131:16116 (type 00, seq 035356, ts 055040, len 000160)
Sent RTP packet to      192.168.3.131:16116 (type 00, seq 035357, ts 055200, len 000160)
Sent RTP packet to      192.168.3.131:16116 (type 00, seq 035358, ts 055360, len 000160)
...

SIP.conf
[6901]
type=peer
username=6901
host=dynamic
secret=6901
qualify=yes
context=webrtc
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
canreinvite=no

You can see from the trace packets that sometimes asteriskgary.local sees no packets from asteriskrtc.local, and at the same time the packets on asteriskrtc.local show half the number of records (there is no "Probation passed - setting RTP source address to 192.168.3.127:15942 which causes twice the number of packets, no idea if this is relevant though).

Please ask if you need anything else. I'm totally stumped with this issue... Note that on asteriskgary.local ICE is not configured, I wouldn't have though it would need it as it isn't talking with the webrtc client itself, it is just talking to an Asterisk server (and that asterisk server is the one which talks to the webrtc client).

Thank you.

Kind Regards,

Gary Shergill

----- Original Message -----
From: "Amit Patkar" <amit at avhan.com>
To: asterisk-users at lists.digium.com
Sent: Wednesday, May 21, 2014 04:41:50 AM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

Hi Gary

You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you 
might have to disable DirectMedia / reInvite for calls going between 2 
asterisk boxes.
I hope this helps to resolve your issue.

*Thanks & Regards,*
Amit Patkar

On 5/21/2014 2:26 PM, Gary Shergill wrote:
> Hi,
>
> I've run into a slight issue when using WebRTC and two Asterisk boxes.
>
> I am using SIPml as the test WebRTC client.
>
> My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local).
>
> Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows.
>
> When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour:
>
> - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901
> - 6901 sees the call and has the option to answer
> - 6901 answers the call
> - 6901 can hear 1000 talking
> - 1000 can not hear 6901
>
> The weird thing is, sometimes it works, sometimes it doesn't...
>
> I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says "Port Unreachable").
>
> Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't.
>
> Thank you for your help.
>
> Kind Regards,
>
> Gary Shergill
>