[asterisk-users] RTP LOG

[Atis Lezdins]

On Sat, Nov 15, 2008 at 6:47 AM, Max Alex <max.asterisk at gmail.com> wrote:
> Hi All,
> Thanks for reply
> i have tried for this, it looks fine for me,
> but is there any way to check rtp log while call is connected or any way to
> enable it to write in log file.
> Please give me some guide lines!
> thanks in advance.

CLI> rtcp stats
CLI> rtcp debug

and as i recall you might also need "sip set debug on" in order to
link this to calls/ip's, as rtcp stats are reporting only SIP call id.

Regards,
Atis

>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen <benny+usenet at amorsen.dk>
> wrote:
>>
>> "Positively Optimistic" <positivelyoptimistic at gmail.com> writes:
>>
>> > exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
>> > exten => h,2,Hangup()
>> > results in....
>> > Set("SIP/rpx2399a-b61fc5e0",
>> >
>> > "CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000")
>>
>> Does it still only report what was in the last incoming RTCP packet?
>>
>>
>> /Benny
>>
>>
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>
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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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