[asterisk-users] Asterisk G722

[zoa]

Asterisk does not support that yet.

Zoa

rachid wrote:
> 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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