[asterisk-dev] why 4 UDP ports per SIP call?

[Klaus Darilion]

Johansson Olle E schrieb:
> 22 nov 2007 kl. 10.29 skrev Klaus Darilion:
> 
>>
>> Olle E Johansson wrote:
>>> 19 nov 2007 kl. 20.53 skrev Klaus Darilion:
>>>
>>>> Hi!
>>>>
>>>> Analyzing Asterisk I see 4 new UPD sockets opened by Asterisk per
>>>> incoming SIP call. I would assume RTP+RTCP=2, but why are there 4
>>>> ports
>>>> opened?
>>> Have you enabled video at all?
>> I have video enabled in Asterisk, but the INVITE sent to Asterisk  
>> had no
>> video description in the SDP. Will Asterisk open the video RTP ports
>> also if the incoming INVITE does not offer video?
> 
> Well, *embarrassing smile*, yes. Hrrm.
> 
> Due to the interesting architecture of the SIP channel, we open all  
> those ports
> for every call, but should close them after the initial INVITE/200 OK/ 
> ACK if I
> remember correctly, or sooner if the actual device doesn't support  
> video.


I am using Asterisk SVN-branch-1.4-r75450M with videosupport=yes.

The incoming INVITE does not include video in the SDP. Also the 200 OK 
from Asterisk does not include video in the SDP. Asterisk opens 4 new 
UDP ports which will be opened for the whole call duration.

Is there a method to find out if the 2 additional ports are really the 
video RT(C)P ports or if they belong to something else?

regards
klaus