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Spark SIP Voice Problem

One Way Voice

We are running 2 servers:

  1. Wildfire Enterprise 3.2.2

  2. Asterisk 1.2.12.1 with about a 12 SIP extensions.

We are running Spark V2.5.0 Beta 2 clients.

The SIP Phone mappings are all working fine, we can register with Asterisk and make calls, but if we make a Spark to Spark voice call, we get one-way sound.

Spark to our Polycom hardware phones, and Spark to X-Lite softphone work fine, with voice both ways. So we know the network and server settings are fine.

Anybody else got one way voice on Spark to Spark calls?

Also…

Also can we have finer detailed control of SIP registration parameters (expiry etc?)

Also call set-up seems to take about 3 seconds before voice is heard. Using X-lite and/or Polycom handset, call setup is nearly instantaneous.

Thanks for any feedback

Liam

Astersik 1.2.2, Wildfire 3.2.0, Spark 2.5.0 Beta 4

I can confirm the Callsetup seems to be taking about 3 seconds or so–at least on my end. This lag in setup time seems to be irrespective of whom i’'m calling. If I call another SIP Phone (polycom) or POTS line via asterisk or even voicemail on asterisk it still exhibits the behavior.

Additionally, the moment I send DTMF from my polycom (not sure why, just did by accident really) it kills audio stream to spark. The connection is still up but spark can’'t hear the remote side.

Iota, thanks for the reply.

Work Around For Call Setup Delay

As an update, I found that the voice delay on call connect can be eliminated on Windows by disabling “NetBIOS over TCP/IP” for all your LAN connections on the PC.

I can confirm that a DTMF tone sent from the Polycom to Spark client kills the RTP stream. I will have a look with Ethereal and see if I can figure out what is happening.

One Way Voice Issue

This issue has been fixed, I believe it was simply a network NAT issue.

Message was edited by: liamgrant

Message was edited by: liamgrant