It will be an issue with the media relay which deduces the IP address of
media based on the initial target of the request. However, I am not
sure if the relay can dynamically adjust sources based on actual RTP
packets sent. Only the author can give a holistic answer to this. If
you claim that RTP is never sent from the caller to the callee based on
wireshark, then the issue is not in the media relay alone since you
would have seen RTP packets come in at least in the network layer and
get lost in the forest of sipX media algorithms.
On 04/11/2011 06:44 AM, Michael Scheidell wrote:
Got an ITSP who sent me instructions, sip trunking, static ip
authentication, with the sip signaling on a different host than the media.
sipx 4.2.0.
first test call was an inbound call from itsp1 to a DID that has call
fwd no answer, call is fwded to a number that matches a dial plan that
is on itsp2,
(itsp1 uses same ip for sip signaling as well as media)
itsp2 is the one with the split sip.. of course, I have never heard of
this.
obviously, the call never completed, not completely.
caller heard ringing, and then 'at the tone, please leave a message'.
but, the external cell phone saw the call coming in.
wireshark shows INBOUND audio from itsp2.s media server, but there is
no outbound rtp to that server (call is an almost answer)
not sure if the split ip (.1 is signaling, .2 is media server) is
supported.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
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