RTP Quality Asterisk 11 vs 13

Hi All,

I have an general question regarding the RTP quality of the Astersik 13. We recently switched our internal PBX to Asterisk 13 and since then we are seeing more Jitter then before.

Yesterday I switched back to Asterisk 11 and the Jitter is gone.

Please note the system is the latestet Asterisk NOW installation and running in a VM. The setting of the VM was and is the same. I just changed the Asterisk version.

Does anyone knows about this topic?

I’m also happy to discuss this in the developer mailing list if you think it is more useful there.

Best regards
Christoph

I’ve seen no reports elsewhere of any problems and haven’t heard (ha) of issues with any deployments I know of. What devices and codecs are in use? Any transcoding going on?

HI Joshua,

thank you for your reply.

I’m also confused.
I’m running the latest AsteriskNOW version with a Asterisk 11.20.
Everything is looking brilliant.
After switching only the Asterisk Version to 13.6 the RTP quality is starting to drop.

I’m working for a company which is developing RTP monitoring software and we are inspecting the inter arrival time of the RTP packets. We are running our own software to monitor our VoIP supplier and there I spotted difference between the Asterisk versions.

I’m happy to provide you more details via webex or so if you are interested as the version switch is just one command.

Please feel free to contract me also directly.

Best regards
Christoph

I prefer to keep all discussions public as Asterisk is an open source project with a community. What exactly are you doing when noticing the change? Just a call between two endpoints? If so we don’t even time in that instance, we just forward traffic.

Hi,

that is also no problem for me.

I’m performing a automated call from the Asterisk via my VoIP provider to a different account on my Asterisk and I can see that the RTP packets of the RTP streams from the Asterisk are effected by more jitter in version 13.6 then in 11.20.
That means I’m analyzing 4 RTP (2 calls) in this setup.

The difference is measured by inspecting the inter arrival time of the RTP streams.

Best regards
Christoph

Hi all,

any ideas how to proceed?

Best regards
Christoph