If you have a remote site using SRST with G711 as the local codec connected to the main Cluster across the WAN
Set up - Only call control, at the moment , is sent across the WAN back to the main Cluster,all PSTN is out local , RTP is only lcoal between the office IPT phones, SRST router has it`s own E1 connection, all handsets use G711, Voice Mail is via a dedicated TDM system. Question -Do you require any additional DSP`s on the actual SRST router for conferencing for example, can the handsets handle it themselves or can they use the conference capability at the main cluster across the WAN- both HW/ and S/W
If you can use the Cluster conference capability - Where do the RTP streams get mixed in the Cluster across teh WAN? Does the mixed RTP stream get sent across the WAN back from the Cluster?
Same set up above but now you can have calls cross the WAN
Let me try to answer you, hope my concept is correct.
If you only allow call control across the WAN, you would need some DSP on the routers to be a hardware conference bridge. This is because the streams are not able to go across the WAN to reach the software conference bridge on the CCM to be mixed together.
In your second scenario, you would be able to use the software conference bridge as the streams are allowed over the WAN.
One thing to note is that software conference bridge only allows G711 conference parties (which is okay since everyone is using G711). Also SW conference bridge only allows up to 6 parties whereas a HW conference bridge can allow up to 8 parties.
Hope i answered your question.
Please correct me if I am wrong in my understanding.
thanks for your response. That makes sense regarding the local DSP as I pursume the "mixing" is done either in the Cluster if software or Hardware but if calls are not allowed across the conference fails.
Do you know a doc which shows how the DSP calculator works on teh cisco web site- a run through?
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