Your user agent (soft or hard phone) does not have either the G.711 mu-law or G.729 codec's enabled. If you wish to configure other codecs then the following order may be considered: If you do enable codecs which are not listed above then keep in mind that this will only work with calls on the Callcentric network or to SIP URIs. If your user agent doesn't support all of these codec's, just skip the ones that it does not have.Īll other codecs should be disabled for compatibility reasons. We recommend that you configure your user agent's codec selection in the following order (with 1 being the highest priority, and 3 the lowest), assuming your user agent supports each of the following. What video codecs does Callcentric support?įor calls from one Callcentric user to another any codec can be used.įor calls to the voicemail system and for error prompting the following codec's can be used:įor calls to the PSTN (traditional landline and mobile phones) the following codec's can be used:įor calls from the PSTN to your Real Phone Number the following codec's will be used: Why can't I always send or receive faxes?
Why?Ĭan I use the Callcentric service with a dialup modem?Ĭan I send and receive faxes with my Callcentric account? When I receive a call my phone rings but when I answer the call it immediately disconnects. To activate the RTCWeb Breaker, the client must include “rtcweb-breaker=yes” as Uri parameter of its AoR (Address of Record).What voice codecs does Callcentric support? The RTCWeb Breaker is disabled by default and it’s up to the client to enable it before registering to the server. For example, FreeSWITCH do not support ICE which means it requires the RTCWeb Breaker in order to be able to connect the browser to a SIP-legacy endpoint.
It’s up to the RTCWeb Breaker to negotiate and convert the media stream to allow these two worlds to interop. PSTN network) do not support these features. The problem is that many SIP-legacy endpoints (e.g. The RTCWeb specifications make support for ICE and DTLS/SRTP mandatory. Webrtc2sip - Smart SIP and Media Gateway for WebRTC endpoints Technical Guide by
0 kommentar(er)
