On 24 August 2012 15:34, Faisal Hanif <[email protected]> wrote: > Steve Davies <[email protected]> wrote: >>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? >> >>Many thanks, >>Steve >> > hi, > > you can simply avoid this by using local ring r option in dial command. > azterisk pass local ring voice to caller and will not bridge b leg audio > until b leg is answered.iin > Regards, > > Faisal Hanif > (sent from phone)
Nice thought, but what if there is a useful reason for the progress audio? If it is sent we want to hono[u]r it, and if it is not, we expect a "180 ringing", and let the SIP device generate the tone, rather than send an unwanted audio stream to use up bandwidth :) For example, some UK ISDN services will give a useful call failure message by sending a progress-frame, followed by some audio. DAHDI and SIP handle this nicely with a 183 progress message, and pass on the message on the un-answered SIP channel. Regards, 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
