Hello, I'm working on request to support SIP trunking with IAD boxes connected to legacy PBXs. Those PBXs are using ISDN for dial-in-only remote management.
For successful management sessions, IADs require so-called Clearmode support (rfc4040). In my target use-case, a management session involves the following devices: Admin PC ---- ISDN modem ---- IAD --- Asterisk ---- IAD --- PBX In this use-case, only Clearmode passthrough is needed as ISDN/Clearmode SIP conversion is done by IAD devices but as Asterisk also supports ISDN, "Clearmode gateway-ing" can help other use cases. >From rfc4040, an example SDP is as follows: audio/clearmode; ptime=10 m=audio 12345 RTP/AVP 97 a=rtpmap:97 CLEARMODE/8000 a=ptime:10 Also from rfc4040: "Clearmode does not use any encoding or decoding. It just provides packetization. Clearmode assumes that the data to be handled is sample oriented with one octet (8bits) per sample. There is no restriction on the number of samples per packet other than the 64 kbyte limit imposed by the IP protocol. The number of samples SHOULD be less than the path maximum transmission unit (MTU) minus combined packet header length. If the environment is expected to have tunnels or security encapsulation as part of operation, the number of samples SHOULD be reduced to allow for the extra header space. The payload packetization/depacketization for Clearmode is similar to the Pulse Code Modulation (PCMU or PCMA) handling described in RFC3551 [5]. Each Clearmode octet SHALL be octet-aligned in an RTP packet. The sign bit of each octet SHALL correspond to the most significant bit of the octet in the RTP packet." 1. Would Clearmode passthough be a welcome addition ? If positive, which are the best stack candidates (pjsip, cna_sip, chan_pri, ...) for it ? 2. Is passthough support much simpler than full support (gateway and passthrough), taking testing into account ? 3. What could be the basics steps for such addition ? Best regards
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