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PBX integration guide – Bandwidth Support Center

    https://support.bandwidth.com/hc/en-us/articles/360021075973-PBX-Integration-Guide#:~:text=The%20following%20ports%20are%20required%20to%20allow%20for,IP%20addresses%20to%20allow%20media%20from%20its%20gateways.
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SIP Port Numbers used by Providers - WhichVoIP

    https://www.whichvoip.com/articles/sip-port-numbers-by-provider.htm
    SIP Control: Port 5060 UDP. Audio (RTP): Ports 10000 to 30000. Ringcentral. Ringcentral port range tends to vary based on the service you purchased but the …

Port Ranges for Supported SIP and VoIP ... - WIN-911 Support

    https://supportdesk.win911.com/support/solutions/articles/24000038761-port-ranges-for-supported-sip-and-voip-providers
    Port ranges for Asterisk: For SIP protocol, open UDP (NOT TCP) port 5060 (SIP) Open ports 10000-20000; Open UDP port 4569 (IAX) Port ranges for sipXecs: SRV records for the SIP communications (port 5060 tcp & udp) SRV …

The World's Audio Network - sip.audio

    https://sip.audio/
    MPEG-4 AAC LD (Direct, ISDN 64 & 128*) MPEG-4 HE-AACv2 (Direct) MPEG-1/2 Layer III (Direct, ISDN 128*) PCM (Direct) Enhanced APT-X (Direct) iLBC (Direct, Relayed) * ISDN & POTS are always relayed, so IP stream must be OPUS, G722, …

What ports should I forward on my NAT device to make SIP ...

    https://support.onsip.com/hc/en-us/articles/203738274-What-ports-should-I-forward-on-my-NAT-device-to-make-SIP-work-
    The default port for udp based SIP signaling is port 5060. Nevertheless, you will still need to check your PBX to find out what port it is using. The RTP media traffic (the actual audio stream) uses a range of udp ports that varies greatly from PBX to PBX and is usually configurable. A typical range might be 10000-20000.

Howto:What Ports are used for Signaling and Voice …

    https://wiki.innovaphone.com/index.php?title=Howto:What_Ports_are_used_for_Signaling_and_Voice_Traffic_in_SIP_and_H.323%3F
    SIP TSIP SIPS H.323 general port range for dynamic ports: 2048 - 59999 by default innovaphone devices use H.245 Tunnelling. This means that H.245 signalling is send via the H.225 connection. If this is disabled or if you use a 3rd party H.323 device, …

Understanding SIP Firewall Configuration - DrVoIP

    https://drvoip.com/support/knowledgebase.php?article=3
    Incoming Ports In a default installation of a VOIP Phone System, the PBX listens for SIP messages on ports 5060 (UDP & TCP). The audio streams are handled by ports UDP RTP ports that you must configure as a range. For example, 9000-9049 (UDP only) – this will allow up to 25 simultaneous calls with VoIP providers or with remote extensions.

Configure your SIP Codec - Knowledge Base - sip.audio

    https://sip.audio/knowledgebase/InQodec_SIP_opus_IP_codec_configuration.php
    AUDIO SETTINGS. From the side menu, click Audio ... Auto Answer: Calls will answer immediately, if this box is checked. LR / LL / RR: Adjust this setting, if you will only use the left or right analog port. Input Gain: Suggested Levels... USB Microphones. Analog Inputs ( XLR SIP Codec ) Blue Yeti: 35%.

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