![]() ![]() The market is expected to reach $23.58 billion in 2026 at a CAGR of 12.4%. The change in growth trend is mainly due to the companies stabilizing their output after catering to the demand that grew exponentially during the COVID-19 pandemic in 2021. The global session initiation protocol (SIP) trunking services marketis expected to grow from $13.03 billion in 2021 to $14.75 billion in 2022 at a compound annual growth rate (CAGR) of 13.2%. , GTT Communications Inc., IntelePeer Inc., Level 3 Communications, Mitel Networks Corporation, Net2Phone, Nextiva Inc., NTT Communications Corporation, Rogers Communications, Tata Communications Ltd., Telstra Inc., Twilio Inc., Vodafone Group PLC, Vonage Holdings Corporation, Voyant Communications LLC., West Corporation, Allstream Inc., Plivo Inc, Ringcentral Inc., Sangoma Technologies Corporation, ShoreTel Inc. 15, 2022 (GLOBE NEWSWIRE) - announces the release of the report "Session Initiation Protocol (SIP) Trunking Services Global Market Report 2022". ![]() , Verizon Communications, 3CX, AT&T Inc, Bandwidth, BT Group plc, CenturyLink, Colt Inc. ![]() Reply to this email directly, view it on GitHub, or mute the thread.Major players in the Session Initiation Protocol (SIP) trunking services market are 8x8 Inc. You are receiving this because you authored the thread. Subject: Re: Putting Session on Hold ( #591) I would try to get the initial renegotiation adding video to stop happening from Asterisk and see if that resolves your issue. From a SIP perspective we are doing all of the correct things with the information provided by the signaling and browser. There is really not much we can do about this in SIP.js. SIP.js then sends the appropriate SIP messages. When the answer SDP comes back for the hold, we pass it to the browser and the browser rejects it. This is another place the browser may not be doing the right thing. Then when you try to place the call on hold, it changes to UDP/TLS/RTP/SAVPF and includes a bunch of codecs that were not originally offered in the initial offer/answer. AFAIK all of this is not supported by Chrome and should be rejected on the spot, but Chrome answers it and allows it to go through. It add MPV (rtp codec 32 which is MPEG) via RTP/AVP (unencrypted RTP). The browser may also be misbehaving, but it is tough to tell.Įssentially what is happening is that you initially set up an audio call. It looks like SIP.js is doing what it is supposed to be doing. I would recommend following up with them. It appears that Asterisk is doing something weird. Observing call to get hold (music on hold). LoggerFactory.print sip.js:586 LoggerFactory.(anonymous function) sip.js:603 Logger.(anonymous function) sip.js:597 setRemoteDescriptionError sip.js:9702 Promise.catch (async) setDescription sip.js:9701 receiveReinviteResponse sip.js:5071 receiveResponse sip.js:3902 InviteClientTransaction.receiveResponse sip.js:2997 UA.onTransportReceiveMsg sip.js:7892 EventEmitter.emit sip.js:743 onMessage sip.js:9003 sip.js:586 Mon 17:36:09 GMT+0500 (Pakistan Standard Time) | sip.inviteservercontext | Could not set the description in 2XX response LoggerFactory.print sip.js:586 LoggerFactory.(anonymous function) sip.js:603 Logger.(anonymous function) sip.js:597 onFailure sip.js:5072 Promise.catch (async) receiveReinviteResponse sip.js:5071 receiveResponse sip.js:3902 InviteClientTransaction.receiveResponse sip.js:2997 UA.onTransportReceiveMsg sip.js:7892 EventEmitter.emit sip.js:743 onMessage sip.js:9003 sip.js:586 DOMException: Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. Sip.js:586 DOMException: Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. ![]() Putting session on Hold is causing call disconnection. ![]()
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