Quantisation of transmission interval using G.723 with PJSIP


I have traced a G.723.1 encoded RTP session between two Asterisk server and noted that the interval between datagrams alternates between 20ms and 40ms while the actual period is 30ms. Asterisk is transcoding between G.711 and G.723.1.

This leads me to think that the Asterisk framework only schedules the generation of RTP audio every 20ms. Is this the case and is it possible to alter this setting so that it could, for example, send out packets every 10ms?