
@Cynthia, I have following this thread. I just had a thought. QoS is recommended for real-time applications. One of the mechanisms is header compression and packet fragmentation (cRTP and LFI). The reason as to why this is done is because VoIP packets are smaller compared to say FTP packets. If FTP and VoIP packets are sent using the same link, the serialisation delay for the FTP packets could be so significant such that the VoIP packets are delayed or droppped. Please remember that the latency from interactive applications such telnet may differ from the real-time latency since the serialisation delay for telnet packets is quite low. Just to paint the picture, the avearge VoIP packet is 40 bytes(IP,UDP,RTP) while the FTP packets could be 150 bytes (IP,TCP,Payload). My recommendation, troubleshoot layer 2 and 3. If it doesn't work, try QoS first. @Michuki, Thanks for info. I have learnt a lot. On Thu, Nov 5, 2009 at 11:06 AM, aki <aki275@googlemail.com> wrote:
@Cyn..th...ia............call drop, engaged tone........ :-)))
As Mich has also pointed out, packet losses could be the main cause. To get away from QOS, jitter and the rest try to get a separate circuit for voip which will eliminate the issues, but first do check the for link drops and packet losses to establish cct parameters. Using QOS and the rest will affect quality at some point eg Monday morning when mail download traffic is high. HTHs.
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