Call flow when creating SIP trunk

[roland.theisen]

Hello,

I have a question regarding the call flow of the phones when using a SIP trunk.

The senario we want to implement is the following:

CIsco IP Phone ---> CUCM ---> SIP Trunk ---> Asterisk ---> Phone

I am aware that the signalling will go the above mentioned way, but what about the RTP stream? Will it go through the SIP trunk or will the two phones communicate directly without the trunk?

If the trunk is used for the RTP stream is there a limitation in geographical distance?

Thanks you for any answer

Roland THEISEN

[Jaime Valencia]

This is Video Over IP, please move to a relevant area.

[EmilTaghiyev]

SIP is for signalization

RTP should be opened between IP Phones (RTP will go directly)

If it is behind NAT U can read some info here

http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html