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rachid at mbdsys.com
Guest
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 Posted: Thu Feb 07, 2008 12:11 pm Post subject: [asterisk-users] Asterisk G722 |
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Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
Below wireshak trace:
<-------------------------------------------------------------------------------------------------------------------->
My Invite:
INVITE sip:600 at asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
Contact: <sip:Manager at 192.168.10.12:5060>
Authorization: Digest username="Manager", realm="asterisk",
nonce="1c8c3fd9", uri="sip:600 at asterisk",
response="5d32f87fa423cd2f1bf9aefb8cf920b6", algorithm=MD5
Max-Forwards: 70
User-Agent: wengo/v1/wengophoneng/wengo/rev54/trunk/
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 365
v=0
o=userX 20000001 20000001 IN IP4 192.168.10.12
s=A call
c=IN IP4 192.168.10.12
t=1202402970 1202406570
m=audio 10600 RTP/AVP 0 8 109 3 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:109 G722/16000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
m=video 10702 RTP/AVP 34 31
a=rtpmap:34 H263/90000/1
a=rtpmap:31 H261/90000/1
<-------------------------------------------------------------------------------------------------------------------->
Asterisk response:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.12:5060;branch=z9hG4bK2600322761;received=77.203.231.140
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>;tag=as5c1447b6
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
User-Agent: Asterisk PBX SVN-trunk-r102777
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:600 at 91.121.31.80:5060>
Content-Type: application/sdp
Content-Length: 397
v=0
o=root 1999706631 1999706631 IN IP4 91.121.31.80
s=Asterisk PBX SVN-trunk-r102777
c=IN IP4 91.121.31.80
b=CT:384
t=0 0
m=audio 18950 RTP/AVP 109 0 8 101
a=rtpmap:109 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18692 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
<--------------------------------------------------------------------------------------------------------------------> |
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zoachien at securax.org
Guest
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| Posted: Thu Feb 07, 2008 12:47 pm Post subject: [asterisk-users] Asterisk G722 |
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Asterisk does not support that yet.
Zoa
rachid wrote:
| Quote: | Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
Below wireshak trace:
<-------------------------------------------------------------------------------------------------------------------->
My Invite:
INVITE sip:600 at asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
Contact: <sip:Manager at 192.168.10.12:5060>
Authorization: Digest username="Manager", realm="asterisk",
nonce="1c8c3fd9", uri="sip:600 at asterisk",
response="5d32f87fa423cd2f1bf9aefb8cf920b6", algorithm=MD5
Max-Forwards: 70
User-Agent: wengo/v1/wengophoneng/wengo/rev54/trunk/
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 365
v=0
o=userX 20000001 20000001 IN IP4 192.168.10.12
s=A call
c=IN IP4 192.168.10.12
t=1202402970 1202406570
m=audio 10600 RTP/AVP 0 8 109 3 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:109 G722/16000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
m=video 10702 RTP/AVP 34 31
a=rtpmap:34 H263/90000/1
a=rtpmap:31 H261/90000/1
<-------------------------------------------------------------------------------------------------------------------->
Asterisk response:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.12:5060;branch=z9hG4bK2600322761;received=77.203.231.140
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>;tag=as5c1447b6
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
User-Agent: Asterisk PBX SVN-trunk-r102777
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:600 at 91.121.31.80:5060>
Content-Type: application/sdp
Content-Length: 397
v=0
o=root 1999706631 1999706631 IN IP4 91.121.31.80
s=Asterisk PBX SVN-trunk-r102777
c=IN IP4 91.121.31.80
b=CT:384
t=0 0
m=audio 18950 RTP/AVP 109 0 8 101
a=rtpmap:109 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18692 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
<-------------------------------------------------------------------------------------------------------------------->
_______________________________________________
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To UNSUBSCRIBE or update options visit:
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kpfleming at digium.com
Guest
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| Posted: Thu Feb 07, 2008 1:37 pm Post subject: [asterisk-users] Asterisk G722 |
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zoa wrote:
| Quote: | Asterisk does not support that yet.
|
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM) |
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digium at sanguinarius...
Guest
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| Posted: Thu Feb 07, 2008 1:42 pm Post subject: [asterisk-users] Asterisk G722 |
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Kevin P. Fleming wrote:
| Quote: | zoa wrote:
| Quote: | Asterisk does not support that yet.
|
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
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Is it (from vague memory) because G.722 @ 16Khz is called G722/8000 ? |
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Guest
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| Posted: Thu Feb 07, 2008 1:55 pm Post subject: [asterisk-users] Asterisk G722 |
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rachid wrote:
| Quote: | Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
|
It's known as 'wideband' audio, to provide you with a keyword you can
use to track asterisk's implementation of it.
From voip-info[1]: "... Speex - which supports 8, 16 and 32 kHz
sample rates and is open source freeware. So if you are looking for
wideband VoIP, look at /Speex/. " and "A caveat for Asterisk hacks:
The internal guts of Asterisk are still substantially geared for 8 kHz
sampling, so arriving wideband signals will end up downsampled. I
understand this is pervasive enough in the core code that it is not
likely to evolve past 8 kHz for some time to come. "
So just tell your client to not /ask/ for 8kHz audio.
I wonder, in a SIP reinvite situation, the UAs would re-choose codecs,
wouldn't they? So even if asterisk didn't support 16kHz for, say, IVRs,
two UAs, once REINVTEd, probably *would* choose 16kHz if they agreed on
it. Am I right on REINVITEs providing opportunity for UAs to battle
out codecs again?
Moj
[1] http://www.voip-info.org/wiki/view/Wideband+VoIP |
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sip at arcdiv.com
Guest
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| Posted: Thu Feb 07, 2008 2:03 pm Post subject: [asterisk-users] Asterisk G722 |
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From the RFC:
"Even though the actual sampling rate for G.722 audio is 16,000 Hz, the
RTP clock rate for the G722 payload format is 8,000 Hz because that
value was erroneously assigned in RFC 1890
<http://www.faqs.org/rfcs/rfc1890.html> and must remain unchanged for
backward compatibility. The octet rate or sample-pair rate is 8,000 Hz."
Thomas Kenyon wrote:
| Quote: | Kevin P. Fleming wrote:
| Quote: | zoa wrote:
| Quote: | Asterisk does not support that yet.
|
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
|
Is it (from vague memory) because G.722 @ 16Khz is called G722/8000 ?
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Guest
|
| Posted: Thu Feb 07, 2008 2:11 pm Post subject: [asterisk-users] Asterisk G722 |
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Mojo with Horan & Company, LLC wrote:
| Quote: | So just tell your client to not /ask/ for 8kHz audio.
|
As Kevin just pointed out, apparently you do NOT have to tell your
client to ask for 8kHz audio. May I ask what client you are using?
Moj |
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