webRTC (sipml5) network problem - no sound

Asterisk Asterisk Support
1

I have two asterisk servers running, with identical asterisk configuration (/etc/asterisk/). One of them works perfectly fine with both Zoiper client(Android) and sipml5 (webRTC). However, the second server has problems with sipml5 even though it has the same asterisk configuration as the first one. Here are the differences:

Server1 (works fine):

  1. Runs on a network that does not have network restrictions other than the device’s own firewall.
  2. Runs on a separate device (CentOS 6.5 on shuttle PC)

Server2 (doesn’t work with sipml5):

  1. Runs on a network with additional firewall in addition to server’s own firewall.
  2. Runs on a VM (CentOS 6.5)

Here are some things that I observed at Server2:

  1. Works fine with Zoiper to Zoiper calls.
  2. When “hello-world” is called from sipml5, there is no sound, even though the log shows that the call is received, “hello-world” is played and then hung up.
  3. When a Zoiper client is called from sipml5:
    • Zoiper rings
    • When answered I can hear an audio played by the server
    • Detect when dtmf is pressed
    • No sound when call is established
    • If sipml5 puts on hold, hold music is heard at the Zoiper client.
  4. When sipml5 is called from Zoiper:
    • sipml5 rings
    • When answered I can NOT hear an audio played by the server (the log shows it is played)
    • No sound when call is established

All of these scenarios work at Server1 without any issues. And I believe this is a network issue at Server2. Server 2 has only public IP (so not behind NAT?).

I called “hello-world” on both servers from sipml5, and here is the link to the log files:
drive.google.com/folderview?id= … sp=sharing

I have no idea how to proceed… I am desperate for a help. Thanks!

2

You don’t have two identical servers. First mistake. You are configuring same things when obviuosly you don’t have.

  1. If you have two firewalls in the second asterisk then configure those firewalls properly.
  2. You already has a working scenario so replicate if you can’t, adapt the rules.
  3. Sound issues are mostly related with nat & stun. Check your rules(and use debugs not just cli verbose).

There is a sticky topic about how to troubleshoot this kind of issues, take a look there. Finally be sure that your VM are in bridged and no NATted mode.

3

[quote=“navaismo”]You don’t have two identical servers. First mistake. You are configuring same things when obviuosly you don’t have.

  1. If you have two firewalls in the second asterisk then configure those firewalls properly.
  2. You already has a working scenario so replicate if you can’t, adapt the rules.
  3. Sound issues are mostly related with nat & stun. Check your rules(and use debugs not just cli verbose).

There is a sticky topic about how to troubleshoot this kind of issues, take a look there. Finally be sure that your VM are in bridged and no NATted mode.[/quote]

if asterisk server and client webrtc together in Lan. I have to establish a connection to turn/stun server or not?

4

Thank you for your reply, and sorry for my late reply. I decided not to use sipml5, and instead I am using CSipSimple :slight_smile: So far I need the client only at the Android side.