Please help: trancoding or not ?

Unanswered Question
Jun 15th, 2009
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Hi,


Have a question about necessity of transcoding resource. Hope someone will help.

Have the scheme attached to case. The settings are the following:


1. MGCP-GW & IP Phone belong to device pool "Site_A_Pool" that is associated with region "Site_A_Region".

2. H323-GW belongs to device pool "Site_B_Pool" that is associated with region "Site_B_Region".

3. Region "Site_A_Region" has G.711 codec for calls withing itself.

4. Region "Site_A_Region" has G.729 codec towards to region "Site_B_Region".

5. Region "Site_B_Region" has G.729 codec towards to region "Site_A_Region".


CALL FLOW:


1. Audio Conferencing Hardware initiates 2 (two) calls at the same time:

towards to IP Phone (DN) and Analog Phone on site B (via Route Pattern).

2. Both calls are connected successfully. IP Phone's user hears voice message from AC Hardware. It's ok.

3. Remote FXS can't hear and cannot be heared. Below is the relevant output from 'show call active voice brief'


{


1996 : 3954 429045680ms.1 +0 pid:0 Originate connecting

dur 00:02:27 tx:5008/78478 rx:4616/58276

IP 10.10.20.1:18834 SRTP: off rtt:58ms pl:0/0ms lost:0/0/0 delay:70/70/70ms g729br8

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

1996 : 3953 429045680ms.2 +10 pid:0 Originate active

dur 00:02:27 tx:4616/95204 rx:5316/127600

Tele 0/1/0:15 (3953) [0/1/0.6] tx:141060/89090/0ms g729br8 noise:-84 acom:6 i/0:-79/-79 dBm


1997 : 3956 429045700ms.1 +0 pid:0 Originate connecting

dur 00:02:27 tx:7264/1162240 rx:7263/1162080

IP 10.10.10.1:20718 SRTP: off rtt:0ms pl:143195/0ms lost:0/1/0 delay:60/60/65ms g711ulaw

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

1997 : 3955 429045700ms.2 +10 pid:0 Originate active

dur 00:02:27 tx:7263/1220184 rx:7361/1177760

Tele 0/1/0:15 (3955) [0/1/0.2] tx:147220/147220/0ms g711ulaw noise:-84 acom:12 i/0:-79/-66 dBm


}


Do I need transcoder here, or not ?

I wonder why IP-Leg for call_id 1996 has 0-value for 'pl:' parameter ("pl:0/0ms") ?


Thank you.




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parshah Mon, 06/15/2009 - 12:38
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  • Cisco Employee,

Hi,


Can you please explain a bit more in detail what is supposed to happen.


1. What is this audio conferencing hardware? Is it an IOS CFB or a 3rd party device.


2. Once it initiates a call is it supposed to patch the 2 endpoints directly or does the RTP from both the endpoints supposed to terminate to this device?


3. The show call active voice brief you have provided, where was it taken from? The MGCP gateway or H.323 gateway?


Thanks,

Tobivan_Helden_2 Mon, 06/15/2009 - 21:07
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Paritosh,


1. 3rd party device (it receives RTP frm both and transmit mixed RTP then towards them).

2. Both RTP streams are terminated on this gateway.

3. The output has been grabbed from MGCP-GW.


The scheme works fine until recently MGCP-GW's IOS has been upgraded to 12.4(25) (SPSERV feature set).


Thank you.


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