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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761 From: Manager <sip:[EMAIL PROTECTED]>;tag=3871604470 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE Contact: <sip:[EMAIL PROTECTED]:5060> Authorization: Digest username="Manager", realm="asterisk", nonce="1c8c3fd9", uri="sip:[EMAIL PROTECTED]", 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:[EMAIL PROTECTED]>;tag=3871604470 To: <sip:[EMAIL PROTECTED]>;tag=as5c1447b6 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX SVN-trunk-r102777 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[EMAIL PROTECTED]: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 <--------------------------------------------------------------------------------------------------------------------> _______________________________________________ -- 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
