It looks like we only have implementations set in multiples of 10ms. You would have to modify code in switch_pcm.c in the loop where we add the implementations (around line 280). Mike On Jun 6, 2013, at 10:52 AM, Steven Ayre <steveayre@gmail.com> wrote:
Maybe. But the ptime will have a minimum value and possibly need to be multiples of a value. The allowable ptimes are codec-dependant. 5ms may well be too low to be allowable.
You can specify it using a @5i suffix in your codec preferences to override the default.
But I would NOT recommend 5ms.... It refers to how long a sample to put in a single RTP frame (packet). Lower values mean you lose less audio data if you drop a packet but also mean you're sending more packets. Every packet has IP + UDP + RTP headers, so has a certain amount of overhead. There's a sweet spot between efficient bandwidth use and not losing too much data if packets are lost. With a 5ms ptime you're going to have a massive amount of overhead which isn't justified - the packets might well contain more header data than audio data. If you're on a slow connection you might even find you start getting packet loss you don't otherwise get because of the extra bandwidth being consumed by the headers.
I would recommend changing the setting on the client instead to use 20ms.
-Steve
On 6 June 2013 12:57, Khue Nguyen Minh <khuenm@vega.com.vn> wrote: Can I config ptime of FS to 5?
2013/6/6 Steven Ayre <steveayre@gmail.com> Tell your SIP client to use a ptime of 20ms
On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn> wrote: Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com> Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:5:64000
PCMU:0:8000:20:64000
Antonio