[Asterisk-Users] Outgoing voice distortion with Unicall

[Rich Adamson]

Carlos Chavez wrote:
> On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote:
>> Hi,
>>
>> I had similar problem and problem was in SIP ATA device (we use Sipura 
>> 2100). They was set from factory to send 30ms voice frame,
>> when we change frame to 20ms everything work perfectly.
>>
> 
> 	Where in the Sipura configuration is that option?  I cannot seem to
> find it.

Login as admin and advanced.

On the SIP tab, under RTP parameters, the entry "RTP Packet Size:".
Default value is .030, change that to .020 (for 20 milliseconds).