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gmaruzz / saraphone

SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc. SaraPhone gets its name from Giovanni's wife, Sara.
Mozilla Public License 2.0
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sip 183 session progress no audio #11

Closed xbipin closed 4 years ago

xbipin commented 4 years ago

most of my upstream gateways send a 183 session progress and send media of custom ringtone which FS passes onto saraphone but the media doesnt start unless the connected is signalled, anyway to get around this in FS or the dialer?

gmaruzz commented 4 years ago

No, early media is impossible with this release. What I do is: FS immediately answer the call from SaraPhone, and pass it any early media from leg B, before bridging. You can have a look at dialplan entries into "resources" directory

xbipin commented 4 years ago

i just noticed this uses a sip.js library thats from 2017, would it be possible to use the latest one?

ok i can also switch to same behavior in FS dialplan but is there any way in FS we can detect this is a wss call and only then answer the call before bridging?