Supported Codecs 3CX Supports the following codecs: G.711a 8000Hz payload type 8 G.711u 8000Hz payload type 0 G.722 8000Hz payload type 9 G.729a 8000Hz payload type 18 GSM-FR 8000Hz payload type 3 iLBC 8000Hz payload type 111 Speex 8000Hz payload type 97 opus 48000Hz palyload type 112 telephone-event 96-127 8000Hz A special note for G.729a, the Media Attribute annexbno must be accepted.Also G.729a does not accept VAD (Voice Activation Detection).
If any of the above does not apply, negotiation o f G.729a will fail. Codec Negotiation Codec Order The codec priority in 3CX can be configured on a per SIP Trunk basis and is done from the Options tab in the SIP Trunks settings, in the Codec Priority section. From here, codecs can be removed, added and have their order changed. It is recommended to remove unnecessary codecs to reduce the MTUpacket size of outgoing INVITE messages. How the order of the codecs affects the codec negotiation will be explained in the following sections. Outgoing Calls When creating the INVITE message for an outgoing call through a SIP Trunk, 3CX will place the codecs in the SDP in the exact order they have been configured in Codec Priority section. If the provider in the 200 OK responds with one or more codecs, 3CX will use the topmost codec found in the 200 OK. Apart from the audio codecs, 3CX will also add the telephone-event 101 codec to the SDP for the transmission of DTMF tones over RFC2833. Note that 3CX does not by default send in-band DTMF tones, it relies on the registered endpoint to do so, so if the endpoint does send in-band DTMF tones, 3CX will relay it. Incoming Calls Upon receiving an INVITE from SIP Trunk provider, 3CX will analyze the order of the codecs in the INVITE and will use the topmost codec in the received INVITE, as long as it is also listed in the available codecs in the SIP Trunk settings. This means that the order of codecs in the SIP Trunk settings does not matter. In regards to DTMF, 3CX accepts all telephone-event from 96-127 for RFC2833. If the provider does not send a telephone-event codec, then 3CX will not accept any RFC2833 DTMF tones, but it will still accept in-band DTMF tones. If the INVITE from the SIP Trunk provider does not contain any codec that 3CX supports, or there is no common codec between those in the INVITE and the ones listed in Codecs in the SIP Trunk settings, then 3CX will reply with a 200 OK with no codecs listed and with SDP Port 0. SDP Connection Address Ports For the SDP connection address, 3CX will decide which IP to use in the same way it decides the IP address in the Contact header for the REGISTER messages as explained in section 3.3.3. In the INVITE and 200 OK messages, first 3CX will check if in the Options tab, there is an IP address entered in the Select which IP to us in Contact and Connection fields option. If this option is set to Use Default Settings then 3CX will use the default Public IP address that it has been configured to use. The Local RTP port 3CX will use on each call is a random port in the range of 9000-10999. There is no limitation however on which port the SIP Trunk providers Media Servers are using. Pjsip Codec Negotiation Password If YouveYou will be redirected to the Customer Portal to sign in or reset your password if youve forgotten it. Click on the button in the email body to verify your email address - (if you can not find it, check your spam folder). By continuing to use our site, you agree to our use of cookies.
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