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


On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,

I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.

Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000]
Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000]
Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000]
Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200]
Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000]
Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000]
Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000]
Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200]
Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000]
Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000]
Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000]
Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200]
Substituting codec PCMU@20i@8000h
No audio codec available
No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
(sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP
Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
(sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep
(sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP
Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION
Responding to INVITE with: 488
sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION

Brs,
Khue Nguyen.


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