[cisco-voip] MTP Required or Not on SIP Trunk to CUPS SIP Proxy?

Paul Dillon pdillon at gmail.com
Fri Mar 28 07:39:27 EDT 2008


Hi There,
I am battling with the situation below.  I hope the details make sense. If
any one has any suggestions or leads or has tried something similar please
let me know.
With Thanks
Paul


**
*MTP Required or Not on SIP Trunk to CUPS SIP Proxy**?*

* *

Call manager 6.0.1





Currently we have  IOS SW MTPs & IOS HW Transcoders in SiteA (2 routers) and
in SiteB (1

Router)



SiteA phones are in the SiteA Device Pool and  SiteA  MRGL and will use
SiteA MTPs



SiteB phones are in the SiteB DP and SiteB MRGL and will use SiteB MTPs



SiteA DP calls to SiteB DP are G729 and vice versa.



We have a SIP Trunk into CUPS SIP Proxy and on to CVP Call Server (SIP Proxy
and CVP local to SiteA). Currently MTP required is selected on Trunk.



SIP Trunk into CCM is in TrunkC DP and SiteA MRGL and will use SiteA  MTPs
(Trunk C Device Pool has region settings of G711 to both SiteA and SiteB
device pool regions)



This is ok for SiteA ingressing calls through the SIP Trunk. However SiteB
calls also ingress to CCM through the same SIP Trunk and are routed to IP
phones in SiteB and will hence use

SiteA DSPs (MTPs) instead of SiteB DSPs (MTPs).



Typically all calls to SiteA extensions through the SIP trunk will have
originated from the PSTN incoming through an IOS gateway in SiteA and
entering CVP system and to SIP trunk on routing to an agent



Typically all calls to SiteB extensions through the SIP trunk will have
originated from the PSTN incoming through an IOS gateway in SiteB and
entering CVP system and to SIP trunk on routing to an agent



Internal IP phone users to also need to be able to call to CVP via the SIP
Trunk









So in the event of MTPs being used across the WAN, what is the bandwidth
utilization

involved?











So if we have MTP required unchecked we run into the following issues:



1. Ext calls at G729 through the SIP trunk to CVP. The call is after playing
prompts etc in CVP vxml rerouted back to an agent extension G729 through SIP
Trunk. The call disconnects



2. Ext calls at G729 through the SIP trunk to CVP. Caller hears the prompts
back as fuzzy noise



3. PSTN caller ingresses to CCM via SIP Trunk and routed at G729 to ext. Ext
places call on hold and cant unhold



4. Same as call 3 above except call to ext is G711 but the result is the
same



5. PSTN caller ingresses to CCM via H323 gateway at G711 to ext. Ext answers
the call and transfers it at g711 through the SIP trunk and the call

fails after a few seconds









Basically I am firstly wondering can the 5 call issues be resolved without
MTP required on SIP Trunk.

If not can I force SiteB calls to use the siteB MTPs and not traverse the
WAN for MTPs (despite the SIP

trunk using the MRGL which contains the SiteA MTPs) and what is the
bandwidth impact of using MTP resources over the WAN
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