StephenC Brekeke Member
Joined: 31 Oct 2006 Posts: 18
|
Posted: Thu Nov 01, 2007 4:09 am Post subject: How to change ptime? |
|
|
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 |
|