02-20-2012 04:59 PM - edited 03-16-2019 09:41 AM
my scenario as follwing
i have integrated my cisco network with core switch 4500 series with access GEPON with core called OLT and End device called ONT which through ip phones connected through as shown in the attached diagram
the issue
the device (ONT) Perform Patting which means that multi ip phone behind the ONT have single ip to register in cisco callmanager which causes some issues with on way communication or sometimes no voice heared
my question is if sip end point is used (ipphone ) can the cisco call manager act as sip proxy which means it can intercept the RTP to differentiate between RTP streams if yes how it would be in the sccp phone be done or i have to use external device like session border controller
i need help the OLT provider simulate this scenario with another sip server which has RTP relay option and it works fine i ask if its exist in cisco call manager or any recommended device to be added to the scenario ........... urgent help will be appreciated
02-21-2012 07:31 AM
What kind of phones are you using? I am not familiar with OLT and dont really understand the benefit of using it vs. registering directly to CUCM. Can you elaborate on your solution?
And the short answer would be that no CUCM would not be able to serve here as a proxy
Chris
02-21-2012 07:56 AM
let me clarify
this project access layer it depends on GEPON Technology as access layer
as follwing the voice vlan in range 172.16.3.0/24
ONT works as fowwing it has WAN interface which take ip from the previous range and all ip phones behind the ont as in the diagram using PAT on the ONT going out to the newtork with the same ip of the wan interface let me give u an example
ONT wan interface ip 172.16.3.7
ipphone 1 (SIP) ip 172.16.3.7
ipphone 2(sip) ip 172.16.3.7
cisco sccp ip 172.16.3.7 because the ont made PATTING to inside ipphones
the issue in the follwing scenario is that call manager cannot differentiate between them ,is it require something to differenate between the ip phones like sip proxy or session border controller ???? or what is the recomended to make or add any device and the GPON provider test this scenario with brekee ip pbx software with feature called RTP relay and he sucesseded in calling each other so in cisco call manager what additional thing missing to do???
02-21-2012 02:15 PM
did recognize the scenario or my point ?
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