SIP: The other party can't hear me but I can hear them when using Asterisk

1

Please read carefully all the question and do not rush to a conclusion.

=== Test environment for point-to-point calls, without Asterisk ===

  • City1: A router with NAT and WiFi and 5060 UDP port forwarded to a machine with MicroSIP. Router has a static external IP address.
  • City1: Android phone 1 with CSipSimple 1.02.03 and "Local" account configured
  • City1: MicroSIP 3.10.1 with no account configured
  • City2: Android phone 2 with CSipSimple 1.02.03 and "Local" account configured, in other city, behind it's own NAT

Test result:

  • calls and text messaging are working between Android1 and MicroSIP, regardless of connection (router's WiFi or cell operator's 3G) used.
  • calls between Android2 and MicroSIP are working.

=== Test environment for calls through Asterisk ===

  • City1: The same router with 5060 UDP port forwarded to Asterisk machine
  • City1: Asterisk NOW 11.9.0, behind the router, with "101", "102" and "103" extensions configured
  • City1: MicroSIP 3.10.1 bound to "103" extension
  • City1: Android phone 1 with CSipSimple 1.02.03 bound to "101" extension
  • City2: Android phone 1 with CSipSimple 1.02.03 bound to "102" extension, in other city, behind it's own NAT

Test result:

  • Text messages aren't working at all.
  • Calls from 101 to 103 are working, regardless of connection (WiFi or 3G) used.
  • Calls from 102 to any other extension are working only partially: 101 or 103 can hear voice from 102, but 102 can't hear any voice, even if calling an "100" extension (Asterisk's voice platform).

This last list item is the main question's topic. Would it be a NAT problem, the P2P environment should also not work, but it does. So there is definitely some problem in Asterisk config.

(Or MicroSIP uses some default STUN and therefore everything works?)

Paul

Posted 2015-06-29T13:41:46.230

Reputation: 579

City2: Android phone 1 with CSipSimple 1.02.03 bound to "102" extension, in other city, behind it's own NAT - this NAT is blocking inbound RTP traffic. Try forwarding/triggering the appropriate ports. – DavidPostill – 2015-06-29T14:02:55.310

@DavidPostill: I wrote "please read carefully ALL the question". Why the P2P configuration works? – Paul – 2015-06-29T14:05:41.940

1<shrug>. Try using a SIP analysis tool to see what the difference is. It will show the call setup and what protocols are being requested together with the ports being used. A good tool will also show the RTP traffic as well and you can see if it being blocked. – DavidPostill – 2015-06-29T14:45:46.323

@DavidPostild: could you recommend a good one? – Paul – 2015-06-29T14:50:03.727

Not really. The only one I'm familiar with is a commercial product (was palladion http://www.voip-info.org/wiki/view/PALLADION, but seems to have changed hands a couple of times, now owned by Oracle), I used it in a previous job as a VoiP support engineer.

– DavidPostill – 2015-06-29T14:56:45.777

No answers