[asterisk-users] Media Proxy Mode in Asterik: SIP and H.323
Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --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
Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and H.323
Bilal, Asterisk is an IP PBX and thus a back-to-back user agent; by default, it will proxy media. The only way to disengage it from the media stream is to use signaling protocol-specific mechanisms to coax the endpoints into talking to each other directly; in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf. The H.323 stack in Asterisk may or may not have a similar option, and it may or may not be compatible with SIP on a signaling level. That's if you're connecting one endpoint that's H.323 and one that's SIP. I imagine there's probably a way to do media stream handoff between two legs that are natively H.323 on both ends. But that's the determinant. Also, remember that even if you hand off the media, Asterisk still stays in the signaling path. This is all the more true if you're using it as a signaling gateway between heterogenous protocols. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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
Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and
Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in a la in the following sentence u wrote it below? in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full proxy (signaling + media) so that will solve miss compatibility issues? By the way: Asterisk allow H.323 to talk with SIP? Regards, --- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Bilal, Asterisk is an IP PBX and thus a back-to-back user agent; by default, it will proxy media. The only way to disengage it from the media stream is to use signaling protocol-specific mechanisms to coax the endpoints into talking to each other directly; in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf. The H.323 stack in Asterisk may or may not have a similar option, and it may or may not be compatible with SIP on a signaling level. That's if you're connecting one endpoint that's H.323 and one that's SIP. I imagine there's probably a way to do media stream handoff between two legs that are natively H.323 on both ends. But that's the determinant. Also, remember that even if you hand off the media, Asterisk still stays in the signaling path. This is all the more true if you're using it as a signaling gateway between heterogenous protocols. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --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
Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and
Bilal, On Fri, 13 Jul 2007, bilal ghayyad wrote: Please explain for me what do u mean exactly in a la in the following sentence u wrote it below? in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf It is an English colloquialism that enjoys wide currency; stolen from French, where it means in the manner of or in the style of, for instance, a la provençale (country style). So, in this case I mean that re-INVITEs are accomplished _by way of_ the 'canreinvite' option. Despite the nomenclature, 'canreinvite' in sip.conf actually means not merely CAN attempt to re-INVITE but WILL attempt to re-INVITE as a point of default behaviour. Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full proxy (signaling + media) so that will solve miss compatibility issues? Well, an H.323 endpoint cannot speak directly to a SIP endpoint without a signaling gateway to convert the messages. If you are referring purely to passing RTP media, I suppose it does not matter, if you can broker that handoff somehow between H.323 and SIP. I am not sure if Asterisk has this capability. By the way: Asterisk allow H.323 to talk with SIP? In principle, yes. Asterisk can act as an H.323 gatekeeper, although not as an endpoint (which is irrelevant to your purpose anyway). By far the easiest and cleanest solution is to run both the signaling and the media through Asterisk, if you're making a SIP - H.323 call. This is the approach least likely to cause any compatibility issues since Asterisk is intermediating in the transaction, and I am not sure if an alternative is even possible. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671___ --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
