cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements
555
Views
20
Helpful
7
Replies
Highlighted
Rising star

CUCM 11.x: Questions regarding software conferencing limitations

Hi everyone,

 

My deployment consists of a Mixed mode cluster of 3 CUCM nodes, each one running CM Service. All my endpoints are VOIP phones.

 

I was hoping someone could elaborate on the use of software conferencing within a CUCM 11.x cluster.

According to the SRND 11.x:

 

The software-based audio conference bridges are provided by the IP Voice Media Streaming Application
on Unified CM. The application must be enabled on each individual node in a cluster. A software unicast
conference bridge is a standard conference mixer that is capable of mixing G.711 audio streams and
Cisco Wideband audio streams. Any combination of Wideband or G.711 a-law and mu-law streams may
be connected to the same conference ...

 

If the Cisco IP Voice Media Streaming Application service
runs on the same server as the Cisco CallManager Service, a software conference should not exceed the
maximum limit of 48 participants.

 

1) Is the maximum limit of 48 participants per node which is running both CM Service and IP Voice Media Streaming? So if my MRG consists of three such servers, then I can support up to 48 * 3 = 144 participants?

 

2) Can the CUCM transcode conferences between G.711, G.729 and G.729a (incoming codec depends on region of phone)?  Would it lower further the suggested maximum limit of 48 participants per node if transcoding is used and if so how can this be calculated?  

 

3) Where can I choose which codecs are supported for the software conference?

 

4) Where can I choose how many participants can be in a conference, and how many conferences can be hosted on any one node?

 

5) Is SIPS with LSC certificates supported if all phones in the software conference are SIP phones? 

 

6) Is SRTP supported if all phones in the software conference are SIP phones? 

 

7) Assuming the participant sizing is acceptable for my environment, and my CPU and RAM resources are also adequate per node, is there any reason why I should use an ISR for a conference bridge rather than CUCM?

 

 

Thank you very much for your time!

Everyone's tags (3)
2 ACCEPTED SOLUTIONS

Accepted Solutions
Highlighted
Hall of Fame Cisco Employee

Re: CUCM 11.x: Questions regarding software conferencing limitations

1) yes

 

2) No transcoding, only G711 and G722

3) You can't

 

4) Adjust the CUCM service parameters for ad-hoc and meet-me, how many conferences is: 48 streams (or whatever value you have configured) / service parameters = max conferences

 

5) & 6) No secure conferencing on SW CFB

 

7) I assume you want secure conferences, and for that you need DSPs.

 

Bear in mind resource allocation is static, so every new conference that is allocated, substracts (based on the service parameters) from the 48 streams (or whatever value you have) until they're depleted.

HTH

java

if this helps, please rate

View solution in original post

Highlighted
VIP Collaborator

Re: CUCM 11.x: Questions regarding software conferencing limitations

1) Is the maximum limit of 48 participants per node which is running both CM Service and IP Voice Media Streaming? So if my MRG consists of three such servers, then I can support up to 48 * 3 = 144 participants?

  • See #4 below with the additional comment that I have never tried to link conferences across servers and I don't know that this is possible.
  • The maximum limit of 48 can be raised on a server if you turn off other IPVMSA Service components on that server such as Music On Hold, Annunciator, IVR, etc.

 

2) Can the CUCM transcode conferences between G.711, G.729 and G.729a (incoming codec depends on region of phone)?  Would it lower further the suggested maximum limit of 48 participants per node if transcoding is used and if so how can this be calculated?

  • The IPVMSA service on CUCM cannot transcode. As the documentation you cited says, "Any combination of Wideband or G.711 a-law and mu-law streams may be connected to the same conference ..."
  • If you need G729 support for conference calls, hardware resources are required.

 

3) Where can I choose which codecs are supported for the software conference?

  • See #2.

 

4) Where can I choose how many participants can be in a conference, and how many conferences can be hosted on any one node?

  • The "48 participants" is something of a misnomer. It should read something like "48 threads/connections" or "48 total participants". The maximum number of participants in ad-hoc and meet-me conferences is, by default, 4. These limits are set in the CallManager Service service parameters Maximum Ad Hoc Conference and Maximum MeetMe Conference. The 48 number would then be divided by the maximum participants per conference to indicate the total number of simultaneous conference calls.
  • It is possible to link ad-hoc conferences via a CallManager Service service parameter Enable Advance Ad-Hoc Conferencing, but is not recommended. However, enabling that parameter also enables the "Join Across Lines" feature, so some organizations do enable it.


5) Is SIPS with LSC certificates supported if all phones in the software conference are SIP phones?

  • I don't know for sure, but I don't believe so. I believe CUCM uses SRTP and not SIPS for internal calls, but don't take just my word for it. Hopefully one of the other helpful folks on this board can answer this.

 

6) Is SRTP supported if all phones in the software conference are SIP phones?

  • I don't believe so. I believe you need hardware resources for that.

 

7) Assuming the participant sizing is acceptable for my environment, and my CPU and RAM resources are also adequate per node, is there any reason why I should use an ISR for a conference bridge rather than CUCM?

  • Other than transcoding requirements, no. Remember, however, that this means any external participants in a software-based conference call will have to join via G711 or wideband so the audio stream would have to come in that way if you don't have a transcoder.

 

I hope this helps. Please let us know if you have additional questions.

 

Maren

(Please mark helpful posts as Helpful below.)

View solution in original post

7 REPLIES 7
Highlighted
Hall of Fame Cisco Employee

Re: CUCM 11.x: Questions regarding software conferencing limitations

1) yes

 

2) No transcoding, only G711 and G722

3) You can't

 

4) Adjust the CUCM service parameters for ad-hoc and meet-me, how many conferences is: 48 streams (or whatever value you have configured) / service parameters = max conferences

 

5) & 6) No secure conferencing on SW CFB

 

7) I assume you want secure conferences, and for that you need DSPs.

 

Bear in mind resource allocation is static, so every new conference that is allocated, substracts (based on the service parameters) from the 48 streams (or whatever value you have) until they're depleted.

HTH

java

if this helps, please rate

View solution in original post

Highlighted
Rising star

Re: CUCM 11.x: Questions regarding software conferencing limitations

Fantastic info, thanks Jaime.

 

Can phones with a secure phone security profiles (SIPS) join a software conference, even if SRTP isn't supported during a conference?

 

Wihth a hardware bridge I can be registered with SIPS to the CUCM cluster even if I'm not in a secure conference.

Highlighted
Hall of Fame Cisco Employee

Re: CUCM 11.x: Questions regarding software conferencing limitations

You can use secure signaling, CUCM will invoke the SW CFB.

For HW CFB, you CUCM will always use SCCP and will do any protocol translation that's necessary.

HTH

java

if this helps, please rate
Highlighted
VIP Collaborator

Re: CUCM 11.x: Questions regarding software conferencing limitations

1) Is the maximum limit of 48 participants per node which is running both CM Service and IP Voice Media Streaming? So if my MRG consists of three such servers, then I can support up to 48 * 3 = 144 participants?

  • See #4 below with the additional comment that I have never tried to link conferences across servers and I don't know that this is possible.
  • The maximum limit of 48 can be raised on a server if you turn off other IPVMSA Service components on that server such as Music On Hold, Annunciator, IVR, etc.

 

2) Can the CUCM transcode conferences between G.711, G.729 and G.729a (incoming codec depends on region of phone)?  Would it lower further the suggested maximum limit of 48 participants per node if transcoding is used and if so how can this be calculated?

  • The IPVMSA service on CUCM cannot transcode. As the documentation you cited says, "Any combination of Wideband or G.711 a-law and mu-law streams may be connected to the same conference ..."
  • If you need G729 support for conference calls, hardware resources are required.

 

3) Where can I choose which codecs are supported for the software conference?

  • See #2.

 

4) Where can I choose how many participants can be in a conference, and how many conferences can be hosted on any one node?

  • The "48 participants" is something of a misnomer. It should read something like "48 threads/connections" or "48 total participants". The maximum number of participants in ad-hoc and meet-me conferences is, by default, 4. These limits are set in the CallManager Service service parameters Maximum Ad Hoc Conference and Maximum MeetMe Conference. The 48 number would then be divided by the maximum participants per conference to indicate the total number of simultaneous conference calls.
  • It is possible to link ad-hoc conferences via a CallManager Service service parameter Enable Advance Ad-Hoc Conferencing, but is not recommended. However, enabling that parameter also enables the "Join Across Lines" feature, so some organizations do enable it.


5) Is SIPS with LSC certificates supported if all phones in the software conference are SIP phones?

  • I don't know for sure, but I don't believe so. I believe CUCM uses SRTP and not SIPS for internal calls, but don't take just my word for it. Hopefully one of the other helpful folks on this board can answer this.

 

6) Is SRTP supported if all phones in the software conference are SIP phones?

  • I don't believe so. I believe you need hardware resources for that.

 

7) Assuming the participant sizing is acceptable for my environment, and my CPU and RAM resources are also adequate per node, is there any reason why I should use an ISR for a conference bridge rather than CUCM?

  • Other than transcoding requirements, no. Remember, however, that this means any external participants in a software-based conference call will have to join via G711 or wideband so the audio stream would have to come in that way if you don't have a transcoder.

 

I hope this helps. Please let us know if you have additional questions.

 

Maren

(Please mark helpful posts as Helpful below.)

View solution in original post

Highlighted
Rising star

Re: CUCM 11.x: Questions regarding software conferencing limitations

Thanks for the info Maren. 

 

If the resources are exhausted for the first node, wouldn't the next node be used for a software conference as per MRGL?

Highlighted
VIP Collaborator

Re: CUCM 11.x: Questions regarding software conferencing limitations

Yes, which means that the total number of participants across your three servers is 144, but that is not the same thing as having 144 participants in a single conference call which is what I thought you were asking about.

 

Maren

Highlighted
Rising star

Re: CUCM 11.x: Questions regarding software conferencing limitations

Alright, thanks for the clarification.

CreatePlease to create content
Content for Community-Ad
Future of Work Virtual Summit Day 5

Cisco COVID-19 Survey