[Asterisk-Dev] RTCP-support

[Dan Evans]

I agree that it sems like NAT could be implicated.  This is what we 
observed in a TCP dump, which I will try to recreate.

Time  Asterisk  (src)(dst)             UA (src)(dst)
   |   RTP -->   10102 5004           recvd
   |   RTP -->   10102 5004           recvd
   |   RTP -->   10102 5004           recvd
   |           ?                  <-- RTCP 5005 10103
   |   RTP -->   10102 5005           recvd
   |   RTP -->   10102 5005           recvd
   |   .
   |   .
   V   .

Dan

Filip Olsson wrote:
> Never heard of this problem.
> 
> Do you mean that the src port(the one on the *-box) is switched to RTP+1 
> or that the destination port(the one * sends _to_) is switched to RTP+1 
> although it's just RTP being set?
> If it is the later it sounds like it has something to do with some NAT 
> setting, try disabling NAT on that peer.
> 
> If you can, post a tcpdump to this bug (#2863) and someone can take a 
> look. If you can't, just post the output from 'rtp debug ip <IP>'.
> 
> //Filip
> 
> 
> Dan Evans wrote:
> 
>> A question regarding this:
>>  
>>
>> We have a user agent that sends RTCP packets, but we had to turn that 
>> off when talking to Asterisk.  We found that Asterisk would shift its 
>> RTP port in midstream to that used by RTCP (RTP+1), after it received 
>> an RTCP packet. Did you find this to be a problem?
>>  
>>
>> Dan Evans
>>  
>>
>> Filip Olsson wrote:
>>  
>>
>>> Hi there,
>>>   
>>> I've written a patch that adds some support for RTCP in Asterisk.
>>>   
>>> Some gateways require that the other end send RTCP-packets during
>>>   
>>> the session or the call will be dropped after a couple of minutes or
>>>   
>>> seconds.
>>>   
>>> This patch has been tested against a number of different SIP
>>>   
>>> useragents and gateways including Ciscos 79xx, AlliedTelesyn RG6xx,
>>>   
>>> AudioCodes MP-series, 42Networks, Granstream, X-Lite/X-Pro, Cisco
>>>   
>>> gateways and VocalTec gateways.
>>>   
>>> Some of those clients/servers require that we send these
>>>   
>>> RTCP-reports, the ones that didn't drop the calls before you applied
>>>   
>>> this patch shouldn't drop them after you apply it either =)
>>>   
>>> If you experience dropped calls at regular intervals this patch might 
>>> be the solution.
>>>   
>>> The patch adds a CLI command to 'debug' RTCP-transmissions so you
>>>   
>>> can see the contents of the reports. For those of you that don't know
>>>   
>>> what they contain, apply the patch and you'll see =)
>>>   
>>> Please apply the patch and test it, you'll find more information at:
>>>   
>>> http://bugs.digium.com/view.php?id=2863
>>>   
>>> There is also a patch available for chan_sip that puts the
>>>   
>>> accumulated stats for the session into the userfield of the CDR upon
>>>   
>>> hangup, it's also in 2863.
>>>   
>>> If you report your results there's a big chance I'll fix them.
>>>   
>>> //Filip
>>>   
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>>
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