Cisco 99xx SIP TCP UDP

[Guilherme Mascarenhas]

Hi all,

I'm experiencing intermittent one way audio in the middle of the calls. Look like it happens more often on WAN calls.

So, I just have some questions about how the IP Phones Cisco 99xx works, If anybody help with that it will be really help on the understanding the current scenario.

1. When the Cisco SIP IP Phone 99xx is configured to use TCP and UDP, I already capture some traces and look all calls use TCP.  When it will use UDP?

2. Does a connection between a Cisco SIP IP Phone and another Cisco SIP IP Phone or gateway can change the ports in the middle of the call?

Any help will be really appreciated.

Regard

[Gordon Ross · Accepted solution]

1. TCP (& possibly UDP) are used for signalling. UDP is used for the audio.

2. In theory, yes, the ports could change. In practise, once the audio stream is established, the ports won't change, unless something happens to the call (e.g. call transfer, on-hold, etc)

If you're experiencing audio issues, start by looking at the call stats. (On a 99xx these are Applications->Admin settings -> Status-> Call Statistics. This will show what the phone thinks is happening to the audio stream.You can verify this by Wiresharking the phone too. If you are having problems with lost packets, large jitter, etc., you need to look at your network. (e.g. QoS policies, etc)

GTG

[Guilherme Mascarenhas]

Thank you,