acappel01 / sipml5

Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
0 stars 0 forks source link

Asterisk on AWS; RTP reciving at client end but no voice #190

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1. Instance is on AWS cloud
2. Installed Asterisk 11.11 on centos 6.5 64 bit 
3. Followed the steps mentioned in 
http://forums.asterisk.org/viewtopic.php?f=1&t=91007
4. For stun used url: 'stun:stun.l.google.com:19302
5. For Turn used numb.viagenie.ca

What is the expected output? What do you see instead?
Background file should be played
no voice

What version of the product are you using? On what operating system?
Asterisk 11.11 on centos 6.5 64 bit
Client machine - Google Chrome - Version 37.0.2062.120 

Please provide any additional information below.
My Asterisk is on AWS cloud with elastic IP
ALL RTP plus SIP ports are open both inbound & outbound
I can see RTP following towards the client system, even webrtc-internal shows 
that RTP recived but no voice.
Same thing works well when installed in local infrastructure.

Original issue reported on code.google.com by tusshar....@gmail.com on 22 Sep 2014 at 7:21

Attachments:

GoogleCodeExporter commented 9 years ago
Hello FRiends,

Do any one have solution for hosting sipml5 on AWS

Regards
Tushar

Original comment by gjain...@gmail.com on 29 Sep 2014 at 7:20