in freepbx GUI when I click on reports and then I click on asterisk info and then on the right I click on peers I can see my sip trunk IP address on 5060 but how can I see what IP address the media is going to
It will be different in every SIP session, but will always be within the RTP range specified in Asterisk SIP Settings.
edit - totally misunderstood what was being asked. I thought OP was asking about media port, not address, hence my nonsense.
when I do asterisk -rvvv from the CLI and I make a call should I be able to see the media IP address of SIP trunk provider?
i dont think you can from the GUI
but try this …
ssh into system
launch sngrep
make call and keep it up
find invite in sngrep
locate the invite from the PBX to the trunk - hit enter
in that above detail function f2 and you see something like i have below detailing the media flow
boom!
you would in sip debug and thats what sngrep is helping with – otherwise you would need to pick apart the SDP … its there
…
v=0
o=Sonus_UAC 603749 963705 IN IP4 67.231.13.113
s=SIP Media Capabilities
c=IN IP4 67.231.13.80
t=0 0
m=audio 54980 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
thank you now I can find the media IP address and test for latency,
my SIP trunk provider tells me that’s sip signaling latency and packet loss doesn’t matter only media, is that true and at what point does it start to matter at what number of milliseconds
i wouldnt say they dont matter - but they wouldnt impact voice quality once the call was set up
things like PDD could certainly be impacted at signaling level
at what number of ms does it start to impact
and what would be the recommended number of ms
really no magic number with respect to signaling latency
in general terms this site has some good info -> https://www.voip-info.org/qos/
to answer more specifically we’d need to know more about the particular issue encountered on a call
thank you very much I really appreciate all the help
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