[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 11:27:48 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:
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Project: Asterisk
Issue ID: 15975
Category: Codecs/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
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 11:27 CDT
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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 :
======================================================================
----------------------------------------------------------------------
(0111441) jehanzeb (reporter) - 2009-09-28 11:27
https://issues.asterisk.org/view.php?id=15975#c111441
----------------------------------------------------------------------
sip history showed the following to channels
Curr. trans. direction: Outgoing
Call-ID:
93486-3463142359-533186 at aosbc1.alwaysongroup.com
Owner channel ID: SIP/84.8.191.13-098b85c0
Our Codec Capability: 8
Non-Codec Capability (DTMF): 1
Their Codec Capability: 264
Joint Codec Capability: 8
Format: 0x8 (alaw)
MaxCallBR: 384 kbps
Theoretical Address: 84.8.191.13:5060
Received Address: 84.8.191.13:5060
SIP Transfer mode: open
NAT Support: RFC3581
Audio IP: 84.8.129.165 (Outside bridge)
Our Tag: as7b5c72f4
Their Tag: 3463142359-533193
SIP User agent:
Peername: nextpoint-sbc
Original uri: sip:07976946209 at 84.8.191.13:5060
Caller-ID: 07976946209
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: Yes
Route: sip:07976946209 at 84.8.191.13:5060;user=phone
DTMF Mode: inband
SIP Options: 100rel timer
aovastest01*CLI> sip show channel
3878e89b7cc795e82474602b1b99ee49 at 84.8.129.188
aovastest01*CLI>
* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 3878e89b7cc795e82474602b1b99ee49 at 84.8.129.188
Owner channel ID: SIP/195.219.133.219-098bd648
Our Codec Capability: 8
Non-Codec Capability (DTMF): 1
Their Codec Capability: 8
Joint Codec Capability: 8
Format: 0x80008 (alaw|h263)
MaxCallBR: 384 kbps
Theoretical Address: 195.219.133.219:5065
Received Address: 195.219.133.219:5065
SIP Transfer mode: open
NAT Support: RFC3581
Audio IP: 10.40.126.198 (Outside bridge)
Our Tag: as5a4ad2e3
Their Tag: e19e202c-1dd1-11b2-b973-b03162323164+e19e202c
SIP User agent:
Username: 02070325205
Peername: 02070325205
Original uri: sip:02070325205 at 195.219.133.219:5065
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: Yes
Route:
sip:02070325205 at 195.219.133.219:5065;transport=udp
DTMF Mode: inband
SIP Options: (none)
the user for the incoming leg is nextpoint-sbc the details of which was
setup as
[nextpoint-sbc]
type=friend
disallow=all ; First disallow all codecs
allow=ulaw,alaw ; Allow codecs in order of preference
autoframing=yes
context=default
trustrpid = yes
;dtmfmode=inband
canreinvite=yes
host=84.8.191.13
port=5060
sip show peer give the following information.
* Name : nextpoint-sbc
Secret : <Not set>
MD5Secret : <Not set>
Context : default
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 : 84.8.191.13
Addr->IP : 84.8.191.13 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : 100rel timer
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: Yes
Status : Unmonitored
Useragent :
Reg. Contact :
one thing i have noticed is that the on the outgoing leg, the peername
doesn not show the name of the peer as sylantro unlike the incoming leg
which identifies the peername correctly as nextpoint-sbc.
Issue History
Date Modified Username Field Change
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2009-09-28 11:27 jehanzeb Note Added: 0111441
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