Configuration problems with 2811 and PVDM2-16

Answered Question
Jun 26th, 2007
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I have a 2811 with PVCM2-16 and I want to configure for voice termination (VIC2-2BRI-NT/TE) and transcoding (SIP G729 to PSTN G.911)


How can I configure DSPs for voice termination and transcoding?. I don't use CCM.


"running-config" and "show voice dsp detail":


+++++

isdn switch-type basic-net3

isdn tei-negotiation first-call

!

voice-card 0

no dspfarm

dsp services dspfarm

!

voice rtp send-recv

!

voice service pots

!

voice service voip

sip

no call service stop

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729br8

!

interface FastEthernet0/0

description LAN maqueta

ip address 192.168.1.253 255.255.255.0

duplex full

speed 100

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface BRI0/0/0

no ip address

isdn switch-type basic-net3

isdn tei-negotiation first-call

isdn point-to-point-setup

isdn incoming-voice voice

!

interface BRI0/0/1

no ip address

shutdown

isdn switch-type basic-net3

isdn point-to-point-setup

!

ip route 0.0.0.0 0.0.0.0 192.168.1.254

!

no ip http server

ip http access-class 23

!

control-plane

!

voice-port 0/0/0

!

voice-port 0/0/1

!

dial-peer voice 1 pots

description *** Llamadas con BRI ***

numbering-type national

service session

destination-pattern 88T

direct-inward-dial

port 0/0/0

!

dial-peer voice 2 voip

description ** LLamadas entrantes desde PSTN hacia SIP XXX

service session

destination-pattern 77T

voice-class codec 1

session protocol sipv2

session target dnsxxxxxxxxx

session transport udp

!

gateway

timer receive-rtp 1200

!

sip-ua

nat symmetric role passive

sip-server dns:xxxxxxxxxxxx

!

+++

*DSP VOICE CHANNELS*

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ======== ========== ===== ======= === == ========= == ==== ============

C5510 001 01 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 02 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 03 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 04 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 05 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 06 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 07 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 08 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 09 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 10 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 11 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 12 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 13 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 14 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 15 None 4.4.24 idle idle 0 0 0 0/0

C5510 001 16 None 4.4.24 idle idle 0 0 0 0/0

*DSP SIGNALING CHANNELS*

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ======== ========== ===== ======= === == ========= == ==== ============

------------------------END OF FLEX VOICE CARD 0 ----------------------------


Please advise.

Correct Answer by paolo bevilacqua about 10 years 3 weeks ago

Hola,


transcoding is when you need to change codec on a RTP session to support e.g. voicemail, conferencing. This happens trasparently, like:


[ip phone/g729]----[transcoding]----[voice mail/g711]


You can terminate G.729 or any other codec, to PSTN, analog or digital, without need for transcoding. This happens since cisco has invented VoIP about 10 years ago :)


Now since you have a cisco router, wanted to let you know, on it you can configure "CME" feature and it will support all the IP phones, cisco or other brand, with features same or superior to the popular sip proxies. And does not require external hardware.


Thanks for the nice rating and good luck!

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paolo bevilacqua Tue, 06/26/2007 - 04:21
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What are you using that requires transcoding? Please include a complete network diagram.

Are the basic calls via ISDN BRI working OK ?

rcastanov Tue, 06/26/2007 - 07:03
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Hi p.bevilacqua and sorry for my poor information,


the network diagram is the next:


IP Phone (G.729) -- SIP proxy (G.729) -- Gateway (2811 + PVDM2-16) -- PTSN (G.711) -- analog phone.


I think the Gateway should implement "voice termination" and "transcoding G.729 - G.711) functions, isn't it?


Now, the basic calls via ISDN BRI are working ok, because IP Phones and SIP proxy are configured with G.711 codecs, but I don't know if DPSs are used in this calls. (IP Phone G.711 -- Gateway (BRI) -- PSTN G.711)


I don't know how DSPs are used, when you want to make voice termination and transcodig. Remember that I haven`t a Cisco Call Manager


thank you for your help,



paolo bevilacqua Tue, 06/26/2007 - 07:51
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Hola,


You don't need transcoding for this. Just make sure that in the incoming DP voip, you either configure G.729, or a voice class that list both G.711 and G.729, so it will work with both codecs without further modifications.


Let us know if it works.


hope this helps, please rate post if it does!

rcastanov Wed, 06/27/2007 - 00:34
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Hi again p.bevilacqua,


sorry for my ignorance about this voice configurations, but I don't know if I understand you...


If I call from IPphone (G.729) to PSTN-phone (G.711) why don't the router do transcoding functions?


The article Cisco Enhanced Conferencing and Transcoding for VoiceGateway Routers (http://www.cisco.com/en/US/partner/products/ps5854/products_qanda_item0900aecd8016c2c7.shtml) says this:

Q. Can any other call agents other than Cisco CallManager use the Cisco Enhanced Conferencing and Transcoding for Voice Gateway Routers feature?

A. Not at this time.


Muchas gracias por tu ayuda (from Spain)

Correct Answer
paolo bevilacqua Wed, 06/27/2007 - 01:37
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Hola,


transcoding is when you need to change codec on a RTP session to support e.g. voicemail, conferencing. This happens trasparently, like:


[ip phone/g729]----[transcoding]----[voice mail/g711]


You can terminate G.729 or any other codec, to PSTN, analog or digital, without need for transcoding. This happens since cisco has invented VoIP about 10 years ago :)


Now since you have a cisco router, wanted to let you know, on it you can configure "CME" feature and it will support all the IP phones, cisco or other brand, with features same or superior to the popular sip proxies. And does not require external hardware.


Thanks for the nice rating and good luck!

rcastanov Wed, 06/27/2007 - 01:56
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Thank you very much


I think today I've learned a little bit from VoIP :)

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