[asterisk-dev] No Audio and ICE negotiation logs

Joshua Colp jcolp at digium.com
Tue Oct 9 06:54:37 CDT 2018


On Mon, Oct 8, 2018, at 5:11 PM, Balraj Singh wrote:
> Hi,
> We have Asterisk Installed on AWS with following configs:
> Asterisk Version: 15.4.0 with NAT and Gateway.
> OS: CentOS.
> 
> The problem we are facing is that we are getting Audio while calling using
> a SIP Phone connected to our Asterisk using our Local IP. But, we are
> facing no Audio issue on both sides when calling from an External IP. By
> External IP I mean, calling from a connection outside of our Infrastructure.
> 
> So for that, we have configured NAT settings and stun server in pjsip.conf
> and rtp.conf respectively. But even then we are facing the same problem, NO
> AUDIO!.
> By turning on rtp debug on we can see the following logs:

I'd suggest getting a packet capture of a failing case and looking at the ICE negotiation itself to understand why it fails. You also need to look at the ICE candidates to understand the possible paths. As well this message really belongs on the asterisk-users list or the community forums[1].

[1] https://community.asterisk.org/

-- 
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org



More information about the asterisk-dev mailing list