This is no hard limit (like with 151 ms delay it´s unusable) but a statement about user acceptance of voice quality. You might get away with 200 ms delay as well.
Technically there is a dejitter buffer, which stores data before sending it as voice through the handset with VoIP. The question is on one hand: how large is the dejitter buffer? You should stay within this limit to avoid voice degradation.
There are however also systems, which dynamically adjust the dejitter buffer. Then in any case it should be below 150 ms - (sum of other delays).
Delays - to name a few - are caused for VoIP by: packetization, serialization, processing, transportation.
So in the end you need to calculate a "delay budget" to figure out what the possibilities are in your environment.
A second comment on your jitter: Did you configure Link Fragmentation and Interleaving (LFI) on your link. This is the technical approach to solve the jitter problem on low speed links. Have a look at f.e. "Reducing Latency and Jitter for Real-Time Traffic Using Multilink PPP Roadmap" at
I would recommend anything in a Campus Lan environment under 5ms over WAN links is a different thing depending on what your use to. I deal with a lot of satellite links so I expect to see quite a bit more two way latency than most people 600ms on average. But when dealing with terestrial links anything under 30 is tolerable.
- generally ping packets are treated as low-priority so try and use two end-points that are reasonably free of congestion before using ping
- if the network is prioritising voice packets based on DSCP EF, set the appropriate DSCP value on your ping packets
If you do the above, you can use pings to get a rough idea of the latency. The size of the packet you use should match the size of your VoIP packets. This depends on the codec used. For example, with G.711, your packets could be;
160 (voice payload)
+ 12 (RTP)
+ 8 (UDP)
+ 20 (IP)
= 200 Bytes.
Alternatively, with G.729, you have:
20 (voice payload)
+ 12 (RTP)
+ 8 (UDP)
+ 20 (IP)
= 60 Bytes.
Once you account for the above, you can certainly use pings. Divide the total round-trip time by 2 to give you an approximate one-way latency.
Pls rate the post if it helps.
The size could be lower again if you are using compressed RTP headers.
Question: Have the phone connected and can ring out-but no sound coming from the handset - analog home phone - phone makes connection and pick up at the other end is ok. Can you help?
Question: Have the phone connected and can ring out-but no sound coming from the handset - analog home phone - phone makes connection and pick up at the other end is ok. Can you help?Answer
iPXE boot in ASR9K
This document applies to NCS5500 and ASR9000 routers and has been verified as such.
traditional ECMP or equal cost multipath loadbalances traffic over a number of available paths towards a destination. When one path fails, the traffic gets r...