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Summary:ASTERISK-14898: Unable to change the packetization settings (ptime) for codecs from default of 20ms
Reporter:Jehanzeb Mansoor (jehanzeb)Labels:
Date Opened:2009-09-28 05:27:01Date Closed:2011-06-07 14:07:24
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Codecs/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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 :
Comments:By: Jehanzeb Mansoor (jehanzeb) 2009-09-28 05:27:49

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@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@84.8.129.188:5060;user=phone>
From: <sip:07976946209@84.8.191.13>;tag=3463120258-769089
P-Asserted-Identity: <sip:7976946209@10.40.126.198;user=phone>
Call-ID: 78779-3463120258-769084@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@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@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@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@84.8.191.13>;tag=3463120258-769089
To: <sip:5205@84.8.129.188:5060;user=phone>
Call-ID: 78779-3463120258-769084@aosbc1.alwaysongroup.com
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205@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@195.219.133.219 SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK5f8fb624;rport
From: "07976946209" <sip:07976946209@84.8.129.188>;tag=as2f22d6dc
To: <sip:02070325205@195.219.133.219>
Contact: <sip:07976946209@84.8.129.188>
Call-ID: 16664f4f3abda365522f09bc3623483a@84.8.129.188
CSeq: 102 INVITE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07976946209" <sip:07976946209@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
By: Leif Madsen (lmadsen) 2009-09-28 09:16:47

Could you provide the console output as well, in addition to the sip history of this call? I can't tell which peer is being used in the outgoing INVITE with this trace. Thanks!
By: Jehanzeb Mansoor (jehanzeb) 2009-09-28 11:27:48

sip history showed the following to channels

 Curr. trans. direction:  Outgoing
 Call-ID:                93486-3463142359-533186@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@84.8.191.13:5060
 Caller-ID:              07976946209
 Need Destroy:           0
 Last Message:           Tx: ACK
 Promiscuous Redir:      Yes
 Route:                  sip:07976946209@84.8.191.13:5060;user=phone
 DTMF Mode:              inband
 SIP Options:            100rel timer


aovastest01*CLI> sip show channel 3878e89b7cc795e82474602b1b99ee49@84.8.129.188
aovastest01*CLI>
 * SIP Call
 Curr. trans. direction:  Outgoing
 Call-ID:                3878e89b7cc795e82474602b1b99ee49@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@195.219.133.219:5065
 Need Destroy:           0
 Last Message:           Tx: ACK
 Promiscuous Redir:      Yes
 Route:                  sip:02070325205@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.
By: Jehanzeb Mansoor (jehanzeb) 2009-10-02 11:30:00

Hi, I have tried setting the codec setting in the [general] portion of the sip.conf file and this seems to do the trick. i get  the required codec packetization rate in the invite for the Outgoing INVITE. i guess my problem isn't why the packetization isn't being set, it to do with getting asterisks to recognize that the outgoing call is going through a peer thats been setup in the sip.conf and not just an unrecognized peer.
By: Leif Madsen (lmadsen) 2009-10-05 11:51:32

How are you calling this peer? Can you show the relevant dialplan portion?
By: Jehanzeb Mansoor (jehanzeb) 2009-10-06 03:58:26

Most certainly

[ Context 'default' created by 'pbx_config' ]
 '_*77XXXXXXXXXXX' => 1. macro(dial-sylantro|${EXTEN:3})           [pbx_config]

[ Context 'macro-dial-sylantro' created by 'pbx_config' ]
 's' =>            1. Dial(SIP/${ARG1}@${sylantro})              [pbx_config]
                   2. Hangup()                                   [pbx_config]


sylantro is defined in the globals as

  show  globals
 
  sylantro=195.219.133.219
  TRUNKMSD=1
  TRUNK=Zap/g2
  IAXINFO=guest
  CONSOLE=Console/dsp
By: Leif Madsen (lmadsen) 2009-10-07 08:58:06

Aha. I think this is a dialplan configuration issue.

Try dialing it as:

Dial(SIP/${ARG1}@sylantro)

Or set the value of your global variable to 'sylantro'

Right now you're circumventing all the information you've applied to the peer sylantro, and just dialing it via the IP address directly. That is why setting it in the [general] section makes this work.