[asterisk-bugs] [Asterisk 0015975]: Unable to change the packetization settings (ptime) for codecs from default of 20ms
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 28 05:27:49 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15975
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Reported By: jehanzeb
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 15975
Category: Codecs/General
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: Older 1.4
JIRA:
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-09-28 05:27 CDT
Last Modified: 2009-09-28 05:27 CDT
======================================================================
Summary: Unable to change the packetization settings (ptime)
for codecs from default of 20ms
Description:
Hi, I am currently running Asterisk version 1.4.21
my problem is that even though i have tried to force outbound calls with a
codec packetization rate of 10ms, or 30ms, asterisk keeps sending the
Invite message with the default ptime of 20ms.
my sip.config file for this peer is
[sylantro]
type=friend
disallow=all ; First disallow all codecs
disallow=gsm
allow=ulaw:10,alaw:30 ; Allow codecs in order of preference
autoframing=yes
context=testcontext
trustrpid = yes
;dtmfmode=inband
canreinvite=yes
host=195.219.133.219
port=5065
sip show peer command shows the following settings
Name : sylantro
Secret : <Not set>
MD5Secret : <Not set>
Context : testcontext
Subscr.Cont. : <Not set>
Language : en
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : Yes
PromiscRedir : Yes
User=Phone : No
Video Support: No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : auto
LastMsg : 0
ToHost : 195.219.133.219
Addr->IP : 195.219.133.219 Port 5065
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:10,alaw:30)
Auto-Framing: Yes
Status : Unmonitored
Useragent :
Reg. Contact :
======================================================================
----------------------------------------------------------------------
(0111406) jehanzeb (reporter) - 2009-09-28 05:27
https://issues.asterisk.org/view.php?id=15975#c111406
----------------------------------------------------------------------
call comes in through one end point with a request for ptime=10 asterisk
server then makes an outbound call towards a second endpoint (sylantro) but
trys to negotiate a ptime of 20ms instead of 30ms as setup on the sip.conf
file and shown as configured for 30ms (alaw)
<--- SIP read from 84.8.191.13:5060 --->
INVITE sip:*7702070325205 at 84.8.129.188;user=phone SIP/2.0
Max-Forwards: 138
Session-Expires: 1800;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:5205 at 84.8.129.188:5060;user=phone>
From: <sip:07976946209 at 84.8.191.13>;tag=3463120258-769089
P-Asserted-Identity: <sip:7976946209 at 10.40.126.198;user=phone>
Call-ID: 78779-3463120258-769084 at aosbc1.alwaysongroup.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP
84.8.191.13:5060;branch=z9hG4bK4c0355aab77aec7a93e6e70a97781718
Contact: <sip:07976946209 at 84.8.191.13:5060;user=phone>
Call-Info:
<sip:84.8.191.13>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 347
v=0
o=aosbc1 2147483647 2147483647 IN IP4 84.8.191.13
s=sip call
c=IN IP4 10.40.126.198
t=0 0
m=audio 40542 RTP/AVP 8 18
a=ptime:10
a=fmtp:18 annexb=no
m=image 40544 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (16 headers 15 lines) ---
Sending to 84.8.191.13 : 5060 (no NAT)
Using INVITE request as basis request -
78779-3463120258-769084 at aosbc1.alwaysongroup.com
Found peer 'nextpoint-sbc'
Found RTP audio format 8
Found RTP audio format 18
[Sep 28 10:38:32] WARNING[13765]: chan_sip.c:5159 process_sdp: Unsupported
SDP media type in offer: image 40544 udptl t38
Peer audio RTP is at port 10.40.126.198:40542
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x108
(alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.40.126.198:40542
Looking for *7702070325205 in default (domain 84.8.129.188)
list_route: hop: <sip:07976946209 at 84.8.191.13:5060;user=phone>
aovastest01*CLI>
<--- Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
84.8.191.13:5060;branch=z9hG4bK4c0355aab77aec7a93e6e70a97781718;received=84.8.191.13
From: <sip:07976946209 at 84.8.191.13>;tag=3463120258-769089
To: <sip:5205 at 84.8.129.188:5060;user=phone>
Call-ID: 78779-3463120258-769084 at aosbc1.alwaysongroup.com
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205 at 84.8.129.188>
Content-Length: 0
<------------>
Audio is at 84.8.129.188 port 12846
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.219.133.219:5060:
INVITE sip:02070325205 at 195.219.133.219 SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK5f8fb624;rport
From: "07976946209" <sip:07976946209 at 84.8.129.188>;tag=as2f22d6dc
To: <sip:02070325205 at 195.219.133.219>
Contact: <sip:07976946209 at 84.8.129.188>
Call-ID: 16664f4f3abda365522f09bc3623483a at 84.8.129.188
CSeq: 102 INVITE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07976946209"
<sip:07976946209 at 84.8.129.188>;privacy=off;screen=no
Date: Mon, 28 Sep 2009 09:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 13733 13733 IN IP4 84.8.129.188
s=session
c=IN IP4 84.8.129.188
t=0 0
m=audio 12846 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Issue History
Date Modified Username Field Change
======================================================================
2009-09-28 05:27 jehanzeb Note Added: 0111406
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