How to change ptime?

[StephenC]

1. Brekeke Product Name and version: PBX 2.0.7.2

2. Java version: Standard Edition 6, v1.6.0

3. OS type and the version: Windows 2003 SP2

4. UA (phone), gateway or other hardware/software involved: Snom 370

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : 3

6. Your problem: How to change ptime?

I am using a phone as an external extension with a 64 upload, 384k download. I am using RTP relay and G.729 over the link.

I am getting problems with delay on the outgoing. I looks like the phone can't get the data out over the uplink fast enough. It builds up over time, so that after a couple of minutes you can have several seconds delay.

The SNOM allows setting of packet size (ptime) up to 60ms, and I have tried with 60ms and 30ms. The invite is sent with this value showing:

Sent to udp:211.121.111.121:5060 at 1/11/2007 10:48:01:836 (1210 bytes):

INVITE sip:102@191.161.221.241;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:1027;branch=z9hG4bK-o7o7eq2gp8dd;rport
From: "Lilly Case" <sip:203@191.161.221.241>;tag=21fg4mrhzy
To: <sip:102@191.161.221.241;user=phone>
Call-ID: 3c2679f01280-8jihce2s6m7y
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:203@192.168.1.11:1027;line=bv7uit1h>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.0.17
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 2010281786 2010281786 IN IP4 192.168.1.11
s=call
c=IN IP4 192.168.1.11
t=0 0
m=audio 54412 RTP/AVP 4 18 3 0 8 9 2 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/BO1SlFbwRE0Sl9NRPFPg7/dSHGUvLNGZ3Pqcm0h
a=rtpmap:4 g723/8000
a=rtpmap:18 g729/8000
a=rtpmap:3 gsm/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=encryption:optional
a=sendrecv

but the reply asserts ptime=20, as follows:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:1027;branch=z9hG4bK-o7o7eq2gp8dd;rport=50398
Record-Route: <sip:211.121.111.121:5060;lr>
From: "Lilly Case" <sip:203@191.161.221.241>;tag=21fg4mrhzy
To: <sip:102@191.161.221.241;user=phone>;tag=1193914100312-19386718
Call-ID: 3c2679f01280-8jihce2s6m7y
CSeq: 1 INVITE
Contact: <sip:102@211.121.111.121:5060>
Allow: INVITE,ACK,BYE,CANCEL,INFO,MESSAGE,REFER,NOTIFY,SUBSCRIBE,UPDATE,PRACK
Require: timer
Supported: timer
Session-Expires: 3600;refresher=uas
Content-Type: application/sdp
Content-Length: 176

v=0
o=SYSTEM 340 1 IN IP4 211.121.111.121
s=-
c=IN IP4 211.121.111.121
t=0 0
m=audio 10362 RTP/AVP 18
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

How do I get the PBX to use a different PTIME? I don't see anything in the manuals or on the admin tool.

Steve

[jelly]

http://www.brekeke-sip.com/wiki/index.php?telepark.wiki=d52bc735
Please check the link, it may be helpful. good luck!

[tnp20]

Are you trying to repacketize the RTP data using ptime ?

One solution I have working is OSS with custom version of RTPProxy which repacketizes G.723 and G.729 payloads in 30ms, 60ms and 90ms payloads to save huge on bandwidth.

Is this what you are trying to do with ptime ?