As Patrick says - try lower - Page 2

veeranagowda Berigi · ‎03-08-2017

Hi Can any one please help me over here, getting huge packet loss for the multi site call, Find the below event logs from the one of the endpoint and let me know how to fix this.

DEVICES ARE INVOLVED HERE ALL ARE CISCO SX20, aLL ARE REGISTERED TO cucm WITH SIP URI.

Endpoint 1 ME1 Details
IP: 172.25.XX.XX
Sip URi: 83650@apcucm.com
software version: TC7.3.7.01c84fd has multisite option


Endpoint 2 HK0 APHQ VICTORIA HARBOUR
IP: 
Sip URi: 87717@apcucm.com
Softeware version: TC7.3.7.01c84fd has multisite option


Endpoint 3 SY1 HARBOUR ROOM
IP: 172.16.XX.XX
Sip URi: 82049@apcucm.com
Softeware version: TC7.3.7.01c84fd do not have Multisite option


2017-03-09T11:08:03.161+08:00 a8 appl[1663]: 629820.65 MainEvents I: ActiveSpeakerIndication(c=21,p=28)
2017-03-09T14:08:06.000+08:00 a8 vcodec: DEC_FSM-0: Still experiencing decode errors (32)
2017-03-09T11:08:16.730+08:00 a8 appl[1663]: 629834.22 CuilApp   User admin(1001) about to execute command '/Audio/Sound/Stop' from .
2017-03-09T14:08:25.000+08:00 a8 vcodec: WARNING: Encoder overproduction (ttvenc_h264_ivahd):
2017-03-09T14:08:25.000+08:00 a8 vcodec:     configured rate 320000, actual rate 486561
2017-03-09T14:08:25.000+08:00 a8 vcodec:     4 frame periods of extra delay would be seen if data paced at configured rate
2017-03-09T14:09:01.000+08:00 a8 vcodec: WARNING: Encoder overproduction (ttvenc_h264_ivahd):
2017-03-09T14:09:01.000+08:00 a8 vcodec:     configured rate 320000, actual rate 539536
2017-03-09T14:09:01.000+08:00 a8 vcodec:     5 frame periods of extra delay would be seen if data paced at configured rate
2017-03-09T11:09:55.908+08:00 a8 appl[1663]: 629933.40 MainEvents I: InputAudioChannelChanged(p=29,gid=514) channels=2 encryption=Off muted=False protocol=Off
2017-03-09T11:09:56.396+08:00 a8 appl[1663]: 629933.88 MainEvents I: InputAudioChannelChanged(p=29,gid=514) channels=1 encryption=Off muted=False protocol=AACLD
2017-03-09T11:10:17.979+08:00 a8 appl[1663]: 629955.47 MainEvents I: ActiveSpeakerReported(c=21,p=29)
2017-03-09T11:10:17.981+08:00 a8 appl[1663]: 629955.47 MainEvents I: ActiveSpeakerIndication(c=21,p=29)
2017-03-09T11:11:13.841+08:00 a8 appl[1663]: 630011.33 MainEvents I: ActiveSpeakerReported(c=21,p=28)
2017-03-09T11:11:13.843+08:00 a8 appl[1663]: 630011.33 MainEvents I: ActiveSpeakerIndication(c=21,p=28)
2017-03-09T14:11:35.000+08:00 a8 vcodec: DEC_FSM-0: No decode errors in 209 s, turning output on again
2017-03-09T14:11:35.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x80000000|Prev. frame lost|
2017-03-09T14:11:35.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x80000000|Prev. frame lost|
2017-03-09T14:11:35.000+08:00 a8 vcodec: DEC_FSM-1: No decode errors in 174 s, turning output on again
2017-03-09T14:11:35.000+08:00 a8 vcodec: DEC_FSM-1: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x82000a00|Concealment Applied|Missing slice|Prev. frame lost|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Decode Err: 0x02000a00|Concealment Applied|Missing slice|
2017-03-09T14:11:42.000+08:00 a8 vcodec: DEC_FSM-0: Continually experiencing decode errors, suppressing further output

Appreciate for your help.

Patrick McCarthy · ‎03-14-2017

Try lowering the bandwidth of the calls. When you are hosting a call on the SX20 you are using more bandwidth than a P2P call. My guess is you simply don't have enough bandwidth to support the calls you are trying to make at the hosting site. For example - if your hosting site has a 1.5M connection to the WAN and your call rate is 768 - one call will work perfectly. Two - you'll get some packet loss (depending on movement in the calls), and three would be horrible. Keep in mind that whatever your call rate is you need to add 20% to allow for IP overhead.

If you drop your call rate and the packet loss goes away bandwidth is undoubtedly your problem. Try 384K.

Ray Boland · ‎03-14-2017

As Patrick says - try lower the call bandwidth so that the sum total bandwidth of the multisite call is the same as the point to point

The multisite hosting endpoint should have bandwidth capacity for both calls reserved, generally it should be call bandwidth+25%. So for a 2mb call your network should have 2.5mb reserved. It's possible to have it configured with lower than 2.5Mb but I'd start from there. You should then be able to make two 1mb calls or one 2mb call

Ray Boland · ‎03-09-2017

Further to what Patrick said, compare the packet loss stats of the multisite call with a point to point call to each site individually as well, and then have the two far sides dial each other.

The recommendation for bandwidth reservation is that you reserve call bandwidth+25% for each video call. For instance, if you want to make 2Mb calls, configure the link so the VC unit gets 2.5Mb bandwidth end to end per call.

Also, packet loss is directional (transmit (TX) and receive (RX). Concentrate on the units that have TX packetloss first. It's also easier to focus on getting two sites working perfectly first, then focus on multisite calls after.

veeranagowda Berigi · ‎03-13-2017

Hi Ray,

Thanks for the reply, We are not observing any packet loss for the point to point call (not changing any bandwidth) 6% packet loss for the multi site call.(am seeing packet loss where the confrerence is hoisted and Tx is 2% and Rx is 6%).

thank you

veera.

Ovindo Prastyo Utomo · ‎03-21-2017

Currently i have similiar issue,

I do have SX80 with speakerTrack CE8.3.1, CUCM 11.5, vTS 4.4(1.19), and conductor 4.3.

In my environment,

packet loss only happen when i do multistream call from SX80 in branches offices,

packet loss amount around 10%-30%.

But when i do call non multistream, the packet loss is 0%.

Both multistream and non multistream use 6Mbps and use 3 SX80s as participant for this test call.

Other thing is packet loss in multistream mode has bigger amount packet loss (20%-30%) only in lowest video resolution,

Stream 1080p = Packet loss 8%
Stream 180p = Packet loss 20%-30%
Btw i'm using "Overlay" for video layout

We already talk to WAN admin and also WAN provider and they said this is the best route that they can give for WAN connection. We don't have any QoS and all duplex is in correct configuration.

Anyone experience same issue like this?

Regards,

Ovindo

Ovindo Prastyo Utomo · ‎03-31-2017

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Michael Randolph · ‎07-26-2017

Have you resolved your issues, Ovindo?

Ovindo Prastyo Utomo · ‎07-26-2017

try disable FEC on SX.

dion_austrade · ‎07-26-2017

Unfortunately there is no way to disable FEC on sx20 ruining CE firmware.. either xconfig or web there is no option..

Getting huge packet loss for multi site call.