On 31 August 2012 07:49, Olle E. Johansson <[email protected]> wrote: > > 24 aug 2012 kl. 16:18 skrev Steve Davies <[email protected]>: > >> Hi SIP Gurus, >> >> I've tried to find the relevant RFCs, but am struggling. I can find >> the odd opinion online, but was wondering if anyone could give a >> definitive answer. >> >> If a SIP call is initiated (INVITE) and receives either a "180 with >> SDP", or a "183 with SDP", then the remote party will start to send >> audio for the inband-ringing. Asterisk then passes this audio, and it >> is correctly heard by the caller. >> >> At present, Asterisk will also start to pass back any handset audio in >> return, in theory allowing a conversation to occur on an unanswered >> channel if an endpoint were designed to allow this (free phonecalls >> here we come!). >> >> My question: >> >> Should: >> 1) Asterisk block outbound audio between the 183 Progress and the 200 >> OK packets? >> 2) Replace any outbound audio with silence? >> 3) Replace outbound audio with a special NULL RTP of some sort? Does that >> exist? >> 4) Allow any audio to be sent regardless? >> >> I have implemented 1) at present on our test rig, but the lack of >> outbound RTP causes issues with firewall state not being set-up to >> allow the inbound audio. I am not sure how hard/easy it would be to do >> 2) as you'd need to create silence of the correct duration to replace >> each audio frame. >> >> Thoughts please? > > First, because of Asterisk's RTP implementation we have to send some RTP > packets at this point. You could mute the calling channel before calling and > unmute the channel at answer if needed, but normally sending audio won't > hurt. A normal user should not be able to send early media on a pstn-like > installation where you bill the calls, so there should be little risc of > two-way conversations before an answer. > > In some cases you have to let the caller send DTMF (the famous fed ex > example) in > early media, so we can't block any media by default in Asterisk. > > Using the "r" option in dial causes a lot of issues, since you can still get > busy or congestion when you have early media, so that is not a good solution. > > /Olle >
Excellent information as always Olle. Many thanks. My intention is to make the early-audio prevention in SIP a little more harsh, such that if SIP receives audio before a 183 or 200 is received, it is dropped. This fixes the case where "useless" early-audio is received from a non-SIP (eg ISDN) technology, and can cause an onward node to auto-enable early audio mode, causing silent ringing and other broken behaviours. Cheers, Steve -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
