Re: [Asterisk-Users] chan_h323: forcing 20ms packetisation
Hi All Is there a better mailing list where I should ask these questions ? Thanks Mike O'Connor wrote: Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323: forcing 20ms packetisation
Is this mike oconnor as in the Australian mick oconnor -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike O'Connor Sent: Wednesday, 20 October 2004 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan_h323: forcing 20ms packetisation Hi All Is there a better mailing list where I should ask these questions ? Thanks Mike O'Connor wrote: Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323: forcing 20ms packetisation
HI Mike, You wouldn't be trying to connect to Comindico in Australia by any chance? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike O'Connor Sent: Monday, 18 October 2004 02:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
