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Transcoding in Lync and CUCM scenario RRS feed

  • Question


  • Hi Guys,

    I am reading the blog below and find some point is quiet confusing,

    blog.unplugthepbx.com/2011/04/11/integrating-microsoft-lync-and-cisco-unified-communication-manager-part-4-remote-site-scenario/

    One of the goals that the author tries to achieve is:

    "A voice call from Lync to Cisco across the WAN should be RTAudio across the WAN and be converted to G711 at the mediation server in the site with the Cisco phone"

    I am wondering the call flow based on it, from the article it suggests:

    Lync (local)-----mediation (local)-----WAN------(remote) mediation(transcodes RTAudio to G711)-------cisco phone (????).


    But when the call comes out of the remote mediation server, how can it reach to the remote cisco phone?


    I myself think the call flow should be like below:

    Lync (local)-----mediation (local)-----WAN------CUCM-----(remote) cisco voice gateway---phone. But if it is the case, where the transcoding happens???

    Can someone please give any advise, thanks.


    Regards

    Jim

     

     

    Tuesday, July 29, 2014 6:38 AM

Answers

  • If all you want to do is to ensure RTAudio is used on the WAN, then go ahead and put mediation servers in each site.  However, if your ultimate goal is to reduce WAN utilization, then there's a better way (but it is relatively complex).  

    You don't really need to deploy mediation servers at each site to get calls to transcode to Cisco.  Lync clients are able to talk directly to Cisco using G.711, avoiding the need for transcoding at all, and you can keep the majority of calls off the WAN by using Media Bypass.

    The way I typically do things with CUCM with remote ISRs is setup trunks to each ISR (one each for Chicago, New York and Detroit), and use media bypass to keep most of your voice traffic off the WAN.

    1. Make sure you've configured all your subnets in Lync and assigned them to sites
    2. Create a PSTN gateway pointing to your CUCM. This will handle signalling.
    3. Create trunks off the PSTN gateway for each of the ISRs, using different ports. Set each trunk's RepresentativeMediaIP to the IP address of the ISR. This ensures that media bypass will work at each site.
    4. In Lync, set the trunks to use Media Bypass
    5. For call routing, you have two options:

    5a) Setup standard voice policies for each location to use the ISR in that location for all calls and assign them to your users (works great if everybody typically stays in one place)

    5b) Use location-based routing to ensure that calls will route out the local ISR no matter where the user is currently sitting.  Fairly easy to setup, but hard to manage and troubleshoot because its all done in Powershell.

    With a bit of fiddling, you can ensure that all PSTN calls and Lync-Cisco calls within the local site will route out the local ISR, avoiding the WAN entirely.  Inter-site routing gets a bit more complicated, but if you really want to minimize bandwidth, you can transcode calls to use G.729 via Cisco for the most efficient bandwidth utilization.  This does come at a cost of increased processor utilization on your ISR and potentially reduced call quality with all the transcoding, however.

    Doing it this way does take more work, but if your ultimate goal is to reduce voice utilization of the WAN, I encourage you to check it out.


    Ken Lasko | Lync MVP | UCKen.blogspot.com | LyncOptimizer.com

    • Marked as answer by Lisa.zheng Thursday, August 7, 2014 5:34 AM
    Thursday, July 31, 2014 2:00 PM
  • Can you create a SIP trunk directly to the Cisco Gateways?  If so, I'd suggest putting a mediation server in Chicago, creating a Trunk in Lync for the Chicago Gateway and allow calls destined for Chicago Cisco phones to take this path. Repeat for New York and Detroit, so they each have their own Mediation server and connect to their local Cisco hardware. That way, a Lync to Cisco call that needed to traverse the WAN would do so in RTAudio, then get converted to G.711 immediately before passing into a local gateway.  Your Lync route patterns will determine which calls get sent to which Cisco gateway or CUCM itself.

    If you can get SIP signaling only to the CUCM, but media directly to the gateways you'd have a similar option. 

    If this is not possible and you can only create a SIP trunk directly to CUCM, you'll want your mediation role in Detroit only as anything else would require multiple WAN traversals and just add latency and overhead.  In this case, you wouldn't want to send anything over the WAN as RTAudio.


    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer". SWC Unified Communications


    Wednesday, July 30, 2014 1:31 PM

All replies

  • You'll only pass through a mediation server on the last hop out of Lync, you'll never go through two nediation servers in a row. So this would never happen:

    Lync (local)-----mediation (local)-----WAN------(remote) mediation(transcodes RTAudio to G711)-------cisco phone.

    The call flow on it's way out of Lync and into the Cisco gateway would travel through the mediation role, who's job it is to transcode whatever Lync was using at the time to G.711 and send it on.

    So, your second diagram is more likely correct if you have your mediation role close to Lync and your CUCM is across the WAN.  Transcoding would happen at that mediation server.


    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer". SWC Unified Communications

    Tuesday, July 29, 2014 2:36 PM
  • Thanks for your reply, Anthony.

    What you meant is something like below:

    we need RTAudio transversing the WAN.  

    Lync (local)--------WAN------CUCM-----WAN------(remote)mediation server--------(remote) cisco voice gateway---cisco phone.

    As it is said, we need RTAudio over the wan and being transcoded at the remote mediation server, after being transcoded to G711, the call will go back to the CUCM in the WAN (??) and comes back to the remote cisco voice gateway to reach the cisco phone. Am I right?

    Thanks

    Jim

    Wednesday, July 30, 2014 1:18 AM
  • In your diagram you have it going into CUCM before the mediation server. It will go Lync to mediation to CUCM to phone typically. Where are your CUCM, gateway, and mediation servers located?

    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer". SWC Unified Communications


    Wednesday, July 30, 2014 1:59 AM
  • Our network is kind of like above, we would like having the RTAudio over the wan, so we are looking for putting a mediation server in chicago and New York office.

    If we have put a mediation server in Chicago, when the RTP stream hit the mediation server in Chicago and be transferred to G711, then it will go back to CUCM and comes back the Chicago voice gateway then goes to the cisco phone? I am confused the media flow now.

    Thanks

    Jim

    Wednesday, July 30, 2014 3:19 AM
  • Can you create a SIP trunk directly to the Cisco Gateways?  If so, I'd suggest putting a mediation server in Chicago, creating a Trunk in Lync for the Chicago Gateway and allow calls destined for Chicago Cisco phones to take this path. Repeat for New York and Detroit, so they each have their own Mediation server and connect to their local Cisco hardware. That way, a Lync to Cisco call that needed to traverse the WAN would do so in RTAudio, then get converted to G.711 immediately before passing into a local gateway.  Your Lync route patterns will determine which calls get sent to which Cisco gateway or CUCM itself.

    If you can get SIP signaling only to the CUCM, but media directly to the gateways you'd have a similar option. 

    If this is not possible and you can only create a SIP trunk directly to CUCM, you'll want your mediation role in Detroit only as anything else would require multiple WAN traversals and just add latency and overhead.  In this case, you wouldn't want to send anything over the WAN as RTAudio.


    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer". SWC Unified Communications


    Wednesday, July 30, 2014 1:31 PM
  • If all you want to do is to ensure RTAudio is used on the WAN, then go ahead and put mediation servers in each site.  However, if your ultimate goal is to reduce WAN utilization, then there's a better way (but it is relatively complex).  

    You don't really need to deploy mediation servers at each site to get calls to transcode to Cisco.  Lync clients are able to talk directly to Cisco using G.711, avoiding the need for transcoding at all, and you can keep the majority of calls off the WAN by using Media Bypass.

    The way I typically do things with CUCM with remote ISRs is setup trunks to each ISR (one each for Chicago, New York and Detroit), and use media bypass to keep most of your voice traffic off the WAN.

    1. Make sure you've configured all your subnets in Lync and assigned them to sites
    2. Create a PSTN gateway pointing to your CUCM. This will handle signalling.
    3. Create trunks off the PSTN gateway for each of the ISRs, using different ports. Set each trunk's RepresentativeMediaIP to the IP address of the ISR. This ensures that media bypass will work at each site.
    4. In Lync, set the trunks to use Media Bypass
    5. For call routing, you have two options:

    5a) Setup standard voice policies for each location to use the ISR in that location for all calls and assign them to your users (works great if everybody typically stays in one place)

    5b) Use location-based routing to ensure that calls will route out the local ISR no matter where the user is currently sitting.  Fairly easy to setup, but hard to manage and troubleshoot because its all done in Powershell.

    With a bit of fiddling, you can ensure that all PSTN calls and Lync-Cisco calls within the local site will route out the local ISR, avoiding the WAN entirely.  Inter-site routing gets a bit more complicated, but if you really want to minimize bandwidth, you can transcode calls to use G.729 via Cisco for the most efficient bandwidth utilization.  This does come at a cost of increased processor utilization on your ISR and potentially reduced call quality with all the transcoding, however.

    Doing it this way does take more work, but if your ultimate goal is to reduce voice utilization of the WAN, I encourage you to check it out.


    Ken Lasko | Lync MVP | UCKen.blogspot.com | LyncOptimizer.com

    • Marked as answer by Lisa.zheng Thursday, August 7, 2014 5:34 AM
    Thursday, July 31, 2014 2:00 PM