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Locate RTP packets loss between sites connected by MPLS

cisco_goh
Level 1
Level 1

We experience poor voice quality between IP phones at 2 sites connected by 2M MPLS link. The call statistics show packets lost (RxLost) but no packets discarded. Voice packets are not prioritised in router due to cost issue, but bandwidth utilisation is below 50%. ICMP and RTP packets have same priority over link but ping tests show no packets at all. Average RTT is 50ms. Interfaces at switchports and router interfaces show no packets drop. How can i identify the source of packet loss?

5 Replies 5

Giuseppe Larosa
Hall of Fame
Hall of Fame

Hello goh,

>> Voice packets are not prioritised in router due to cost issue

??

take time to configure LLQ to protect voip traffic on both ends.

I guess you haven't bought a QoS service from the provider but you should what you can on your side.

>> How can i identify the source of packet loss?

it may be in the provider network

>> ICMP and RTP packets have same priority over link but ping tests show no packets at all.

do you mean there is no connectivity between the two phones?

if so check your routing one site may not be able to send back icmp echo replies.

You may need to work with provider stuff on this.

Hope to help

Giuseppe

Hi

Thanks for your reply.

>> Voice packets are not prioritised in router due to cost issue

we have only 16kbps subscribed for priority bandwidth which is insufficient for our volume of call sessions, it will be costly to increase the bandwidth subscription, hence voice is not treated as high priority in MPLS.

>> ICMP and RTP packets have same priority over link but ping tests show no packets at all.

sorry, it should be no ICMP packets LOSS at all. there is connectivity between the phones.

Provider is telling me that ICMP and RTP have same priority in the circuit, since ICMP ping tests show no packets loss, from network perspective there is no issue. However, our ip phone call statistics show RTP lost, i am not sure how i can determine if the loss occurs in the MPLS network.

Hello Goh,

ip connectivity is fine and this is good news.

about RTP losts:

try to get statistics on delay and jitter: there are chances that packets with excessive delay are ignored by the codec : they are arrived too late to be used to rebuild the speech.

This can be an effect of the missing LLQ.

another possibility is that some RTP packets are dropped within provider network but it is difficult to demonstrate this.

Hope to help

Giuseppe

Hi Giuseppe

The call statistics show no pkts discarded, only lost packets. Jitter is not an issue as the avg and max jitter values were monitored and accepatble.

I will look into your suggestion on LLQ, if you know a good link, pls let me know as i am not familiar with it.

Hopefully, the problem is not in the provider network.

Thanks again.

Mohamed Sobair
Level 7
Level 7

Hi Goh,

According to the QoS guide, To have good quality of Voice the following should apply:

1- End to End delay should be lesser than 150ms.

2- Jitter (variation of delay) should not exceed 30ms.

3- Total Drops on the packet should not be more than 1%.

If you experience Packet drops, then it will definetely effect your Voice quality.

To over come the issue, you can implement LLQ which gurantees your VOIP delay and bandwidth.

The link bellow is very useful to view, it describes LLQ and CBWFQ concept , config & montioring.

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/qos/congestion-management-configuring-cbwfq-and-llq-2/player.html

HTH,

Mohamed

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