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New Member

CISCO and Polycom integration via CUBE

Hey Guys,

I believe this one is a good one.

I'm trying to integrate CISCO and Polycom using a CUBE.

What i got until now is:

1. Calls from CISCO to Polycom are perfectly fine.

2. Calls from Polycom to CISCO needs the intervention of a field engineer to set the correct bitrate bandwidth in order to function.

Item 2 is a big problem since endusers are terrible to understand voice/video concepts.

Also, some Polycom endpoints only works with maximum 64 kbps bitrate. No audio is being stablished on calls with bitrates greater than 64...

I want to have this to work automatically. Without user intervention.

Follow bellow my CUBE configuration.

Really appreciate some help.

7 REPLIES
Silver

CISCO and Polycom integration via CUBE

I'm confused on what your actual call flow is?

Something like Cisco IP Phone<->SCCP<->CUCM<->SIP<->CUBE<->Polycom SIP Phone ?

Just mentioning one side is Cisco and one side is Polycom is not enough details as both companies have many different products.

Also, can you collect a "debug ccsip messages" for both types of calls?

VIP Super Bronze

CISCO and Polycom integration via CUBE

Can you send us a debug ccsip messages from the cube?

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: CISCO and Polycom integration via CUBE

Hey Guys,

Here is attached the debug ccsip all:

1. First call we were using 128 kbps and didn't worked.

2. Second call we have used 64 kbps and it worked.

Caller: 93501 - Polycom video endpoint HDX4500

Called: 9219357493 - cisco 6941

and also the CUBE config.

      

The interesting lines: (entire config is attached)

! Config to be applied on both CUBEs:

! 172.23.124.178
! 172.23.124.241

! Voice translation-rules that will allow only authorized traffic on the CUBEs.

voice translation-rule 10
rule 1 /^921935\(....\)/ /921935\1/ -----> rule for Bay View
rule 2 /^921934\(....\)/ /921823\1/ -----> rule for Cittá
rule 3 /^921933\(....\)/ /921821\1/ -----> rule for EBM
rule 4 /^9\(.......\)/ // ----> exclusion this won't match anything. Reason is that i received a request from Architecture to route calls that are only allowed and restrict others.

voice translation-profile VIDEOINTEGRATION
translate called 10

! Check URI for incomming ip address and if match consider only the dial-peer that has the translation-profile


voice class uri 1001 sip
host ipv4:172.23.182.24   -----------> DMA IP Address


! Inbound dial-peer that has the URI command and the translation profile

dial-peer voice 5003 voip

description VIDEOINTEGRATION

translation-profile outgoing VIDEOINTEGRATION

session protocol sipv2

session transport tcp

incoming called-number .

incoming uri via 1001

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec transparent

no vad


! Create a dial-peer that has a pattern 921

dial-peer voice 401 voip
description VIDEOINTEGRATION SEND TO BAYVIEW
preference 1
max-conn 10
destination-pattern 921.......
session protocol sipv2
session target ipv4:172.24.96.98
dtmf-relay rtp-nte

codec transparent

dial-peer voice 402 voip
description VIDEOINTEGRATION SEND TO BAYVIEW
preference 2
max-conn 10
destination-pattern 921.......
session protocol sipv2
session target ipv4:172.24.96.99
dtmf-relay rtp-nte

codec transparent


! Outgoing dial-peer that has the route to 976.......

dial-peer voice 801 voip

description to VIDEOPOC-TO-BAYVIEW

destination-pattern 976.......

session protocol sipv2

session target ipv4:172.23.182.24

dtmf-relay rtp-nte

codec transparent

! Stuff necessary to have compatibility with non CISCO devices.

voice service voip

  sip

    asymmetric payload full

VIP Super Bronze

CISCO and Polycom integration via CUBE

From the logs,

CUBE doesnt seem to like the media types advertised in the media lines for "m=application" in m-line 3 and 4.

m=application 49190 RTP/SAVP 100

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yEx5CIHH6c1lsD771ocBmcdgtHSYuuALrs4jH6Vs|2^31

a=crypto:2 AES_CM_256_HMAC_SHA1_80 inline:rmsGLdS7kDcYS0e3A0E1RtBnf/4NAFc7NVs1vGLkQnFzKE2Erx6yg2hiHpeDCw==|2^31

a=rtpmap:100 H224/4800

a=sendrecv

m=application 40157 UDP/BFCP *

a=floorctrl:c-s

a=setup:actpass

a=connection:new

ct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/sipSPIValidateConnectionAddress: Dest port = 49190

SIP: (2022) Attribute mid, level 3 instance 1 not found.

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/sipSPIValidateStreamAddrType: stream:3, Mode : 1

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 3 = 172.23.124.186

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Error/sipSPIDoMediaNegotiation: Ignoring unsupported media type (2) in media line 3

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/sipSPIValidateConnectionAddress: Dest port = 40157

BRVIX5VALEVG003#SIP: (2022) Attribute mid, level 4 instance 1 not found.

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/sipSPIValidateStreamAddrType: stream:4, Mode : 1

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 4 = 172.23.124.186

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Error/sipSPIDoMediaNegotiation: Ignoring unsupported media type (2) in media line 4

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Error/sipSPIDoMediaNegotiation:

no valid fax or audio streams

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Error/sipSPIHandleInviteMedia: Media Negotiation failed for an incoming call

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:65, category:278

Oct 16 20:33:02 UTC: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[2022], src[6]

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_UNACCEPTABLE_MEDIA_ERR

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Error/sipSPIContinueNewMsgInvite: Unacceptable media indicated for INVITE

Oct 16 20:33:02 UTC: //2022/FD02230288A1/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:65, category:278

The issue lies here..

Can you post a trace of a working call? When you say you set the bit rate to 64k. What do you mean? Is that the payload used on a video codec or ???

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

CISCO and Polycom integration via CUBE

on the same trace there's a call which is working.

same caller and called number.

On the polycom codec there's some settings where you can specify which bandwidth to use on the call. When i do set this to 64 kbps the call completes perfectly fine. audio both sides!

have tried diferent setups on the incomming dial-peer but till now nothing...... i'm bit frustated.

=/

VIP Super Bronze

CISCO and Polycom integration via CUBE

The trace for the working call has only one m=application line and doesnt have the m-line with generic BFCP..

m=application 49196 RTP/SAVP 100

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WiPpzgkcagvZtG8Y7s30ApFqzafkY5C2jKvKHrS5|2^31

a=crypto:2 AES_CM_256_HMAC_SHA1_80 inline:fW10KOrCIDb95XDvcu9ORAqjTXasCR32CFZS/6x2q3TETa3wH1DVzbdc7C9LFw==|2^31

a=rtpmap:100 H224/4800

a=sendrecv

Thats all the application m-lines it has. The failed call has two m-lines for application

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

CISCO and Polycom integration via CUBE

Did you try using content Pass-through?

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1376226

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
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