Is it possible to transfer audio in raw PCM in SoftPhone TS ? and SoftPhone TLS connection getting closed after 2mins of connection, Why ?

Question

Hi folks!In RingCentral SoftPhone TS: https://github.com/ringcentral/ringcentral-softphone-tsI see we can transfer audio on Opus and PCMU, but is it also possible to transfer audio on raw PCM ?I know transferring decompressed audio would be efficient but I’m having to pass the audio to another service in raw PCM and also receiving audio in raw PCM from there, and in the middle I’m having to decode/encode the audio which is causing to high CPU usage, so I was looking if it’s possible to transfer audio in just raw PCM with RingCentral SoftPhone TS as well so I wouldn’t have to decode/encode the audio. Also in SoftPhone TS, I noticed TLS connection for transferring SIP messages is getting closed immediately after 2mins of connection established, so when call continuous more than 2 minutes, we’re not receiving any messages from TLS at all, especially when callee ends to call, we’re not receiving “BYE” message which is causing to have our call active forever, is there any possibility to make the connection expiration time longer ?I saw this line “Expires: 3600”: https://github.com/ringcentral/ringcentral-softphone-ts/blob/e9b7493353e7a0e9444c053c3e00c8910eebf333/src/index.ts#L106Tried to change its value but it didn’t change anything at all, always kept getting the connection closed immediately after 2mins

Tyler850957020 · Answer

I believe the second issue has been fixed here in May:https://github.com/ringcentral/ringcentral-softphone-ts/commit/c8af429182bcd1f100614656f6bb83e0c38177b7Please try the latest version.For now we only support 3 codec: PCMU/8000, OPUS/48000/2 and OPUS/16000.I don’t think uncompress audio is supported. The SDK could support it, but RingCentral SIP server doesn’t support it, so, it won’t work.To list all the codecs that RingCentral SIP server supports, you can make an inbound call to the softphone SDK and watch the inbound INVITE message. Here is the list as I tested:a=rtpmap:0 pcmu/8000a=rtpmap:9 g722/8000a=rtpmap:18 g729/8000a=fmtp:18 annexb=noa=rtpmap:96 ilbc/8000a=fmtp:96 mode=20a=rtpmap:8 pcma/8000a=rtpmap:109 OPUS/16000a=fmtp:109 useinbandfec=1;usedtx=0;ext32-dred-duration=1000;sprop-ext32-dred-duration=1000a=rtcp-fb:109 ccm tmmbra=rtpmap:111 OPUS/48000/2a=fmtp:111 useinbandfec=1;usedtx=0;ext32-dred-duration=1000;sprop-ext32-dred-duration=1000a=rtcp-fb:111 ccm tmmbra=rtpmap:101 telephone-event/8000a=fmtp:101 0-15

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