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VoIP - Technical Information: Difference between revisions
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*Only RTP audio using a-law 20ms is supported. This is generally compatible with all carriers and devices and provides high quality audio.
*Out of band DTMF is accepted using SIP INFO or RFC2833. DMTF can be sent using RFC2833 or generated a-law in-band audio.
*SIP address: Incoming calls can be made from the internet to sip:number@aa.org.uk and delivered just like a normal incoming call. The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted. [Creating VoIP - Calling With a SIP URI|Read more]
*As the system uses phone numbers the domain part is not relevant for registration and incoming SIP calls, but we recommend using the full E.164 number as the username part and aa.org.uk as the hostname, e.g. sip:+443333400200@aa.org.uk. We also accept calls to tel: URIs.
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