If you are talking about the voice stream packets, then they use Real Time Protocal (RTP) which use UDP ports. RTP does not provide mechanisms to ensure timely delivery out of order delivery is still possible, and flow and congestion control are not directly supported. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTP Control Protocol (RTCP) provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate.
So in essence the packets can be delivered out of order (jitter) resulting in bad voice quality such as choppy or garbbled voice.
for more info check out
http://www.networksorcery.com/enp/protocol/rtp.htm
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Chris
