[Freeswitch-users] Dest./port unreachable for RTCP and first few RTP packages

[René Weiss]

Am 09.02.17 um 18:12 schrieb Michael Jerris:
>> - All RTCP packages
>
> We only by default enable rtcp if negotiated in the sdp.  There is a var you can set to override this and use by default even if not negotiated.  Note this may not work in some NAT scenarios.

What var would this be?

Here's the SDP from one of calls I have a trace of:

"INVITE" from us:

v=0
o=FreeSWITCH 1486572028 1486572029 IN IP4 192.168.210.7
s=FreeSWITCH
c=IN IP4 192.168.210.7
t=0 0
m=audio 60306 RTP/AVP 8 0 3 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20


"200 OK" from the other side:

v=0
o=oltszscp-isbc02 293695488 1486632335 IN IP4 138.187.57.135
s=sip call
e=unknown at invalid.net
c=IN IP4 138.187.57.135
t=0 0
m=audio 20052 RTP/AVP 8 0 13 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:on - - - -
a=ptime:20


There's no "a=rtcp:<PORTNUMBER>" attribute, but if I understand the 
information I found correctly then it should not be needed because the 
other side sends the RTCP packages to port 60307 which is next the 
higher port from port 60306 in "m=audio 60306 RTP/AVP 8 0 3 101 13".

Is this correct?

>>
>> - At least in one case the first few RTP packages in a new call (6
>> packages over around 0.12s, after that RTP worked)
>
> This can be normal where we start sending media before the other side has set up their listener, or in nat scenarios, before they have started to send us media.

In this case it's our side who isn't ready to accept the first few
RTP packages send to us.

>>
>>
>> So my questions are:
>>
>> - Should I be worried about these?
>
> No
>
>>
>> - What's the deal with RTCP packages? Can/should I simply ignore them
>>   or is there an option in freeswitch that I should set to handle them?
>
> This all depends on if you want the information from rtcp or not.

I don't think that I need the information.

It's just that I want to have our side of the communication with our 
upstream provider as clean as possible, so that it's easier to identify 
the real issues when we have to debug VOIP problems.