Not getting RTP(UDP) port when using softphone calling(One way audio in softphone calling in citrix)

Question

One of our customer they have installed our application in citrix xendesktop setup(OS-windows server 2016) and they are getting RTP port busy error when softphone calling. we have tried to configure 3rd party softphone. But the same issue. As per our analysis it seems the OS issue.

Issue details-

APP requests a UDP socket with a local port of XX

Windows says okay and gives the UDP socket but with port XY

APP now sends an INVITE with SDP containing XX as a destination port for RTP

It receives an OK with SDP where the port for media on the other participant is YY

So now a call is established

APP sends media from port XY to YY

so other participant is able to hear, the other end sends on xx

but other participant sends RTP from YY to XX which goes nowhere . It something RTP never return port which was requested.

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So can you please help us, how and what configuration will do, so that we can resolve this issue.

Citrix version - 7.18 OS- windows server 2016 client os- win10

Answer 1

Hello

Thank you for your question and reaching out.

A "port" is a standardized channel on a router that allows you to receive traffic from other internet users. There are 65535 ports on a traditional router. Many ports are assigned for specific traffic protocols. For instance, HTTP traffic comes through port 80. SIP traffic comes through port 5060. RTP traffic varies between phone systems, but a typical range might be 10000-20000.

If your router or computer is using NAT (Network Address Translation) or a firewall, these features might close SIP and RTP ports so that packets never reach your phone. When the proper ports are not forwarded or opened, your calls could drop altogether or fail to initiate.

SIP is only a signaling protocol – it doesn’t actually carry the voice of a telephone conversation. To transfer voice between VoIP endpoints, SIP works in tandem with other protocols that transmit the voice information as payload. These include Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP), both of which are User Datagram Protocol (UDP)-based protocols. This means that SIP message exchange and voice packet exchange occur over two separate sessions, or channels.

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