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You may get problems that audio sounds broken, this is generally caused by out of sequence packets.

With most systems this can be fixed by enabeling a jitter buffer

With FreePBX (Asterisk based Distro's like Trixbox, Elastix etc) you can change this in the sip_general_custom.conf

here are the variables you will require:

;Enable Jitter Settings
jitterbuffers=4
jbenable=yes
jbforce=yes
jbimpl=fixed
;jbimpl=adaptive
jbmaxsize=50
jbresyncthreshold=100
jblog=yes

Asterisk sip jbenable = yes|no
Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4)

Asterisk sip jbforce = yes|no
Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no". (Added in Version 1.4)

Asterisk sip jbmaxsize = Number
Max length of the jitterbuffer in milliseconds. (Added in Version 1.4)

Asterisk sip jbresyncthreshold = Number
Jump in the frame timestamps over which the jitterbuffer is resynchronized. Useful to improve the quality of the voice, with big jumps in/broken timestamps, usually sent from exotic devices and programs. Defaults to 1000. (Added in Version 1.4)

Asterisk sip jbimpl = fixed|adaptive
Jitterbuffer implementation, used on the receiving side of a SIP channel. Two implementations are currently available - "fixed" (with size always equals to jbmaxsize) and "adaptive" (with variable size). Defaults to fixed. (Added in Version 1.4)

Asterisk sip jblog = no|yes
Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4)

If you have any problems please Raise A Ticket