>There is really no reason in > most cases to pass the media through an asterisk box, unless you need to > transcode it. >>Hi
>>Sorry for the newb question but what exactly do you mean by this? The "media stream" (or "media path") refers to the audio passed between phones, Asterisk, and a service. This can be handled in two fashions: 1. Asterisk acts as the "broker" between two devices (or in this case, a device and a service) handling call setup and teardown. Once the call is set up, Asterisk "steps out of the way" and allows the two devices to talk to each other directly, without having to relay the media stream. This lessens load on the Asterisk server and leaves the media stream cleaner, since there are no artifacts from Asterisk re-processing the audio (an artifact could be defined as network delay, sound distortion from re-converting the audio from one format to another, or jitter from audio packets arriving early, late, out-of-order, or not at all) 2. Asterisk stays in the media stream, retransmitting the audio from the source to the destination as essentially two calls, from the perspective of Asterisk. This is useful in the case of the source device not transmitting audio in a format that the destination device (or service) expects. Asterisk converts the audio in the format desired on-the-fly. Staying in the media stream can introduce artifacts in the audio as detailed in 1) so to transmit audio in the cleanest fashion possible, it is desirable to have the call endpoints talk direct to each other as much as possible. In Asterisk, this behavior is controlled using the "canreinvite" parameter in sip.conf and also by use of the "T" parameter in the Dial() application. Using the "T" parameter will leave Asterisk in the media stream, since T controls whether or not Asterisk will listen for a DTMF tone to initiate a call transfer. A useful application for leaving Asterisk in the media stream, which I do in my installation, is so that Asterisk can record the calls. Obviously, if the two devices are talking directly to each other, Asterisk cannot record the call. hth _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz