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ffacilities
Beginner

IAD 243x as SIP <> ISDN gateway with iLBC

Hi all,

I have a couple of IAD 243x (specifically IAD2431 and IAD2432) running IOS 15.1(4)M5 that I'm using as SIP to E1 PRI ISDN gateways. They're configured with one POTS dial-peer for the ISDN and a couple of VoIP dial-peers depending on the incoming called number from the ISDN. This has been working fine with G.711 A-law end-to-end, but now I'm trying to get it to work with iLBC and having problems.

My dial-peer configuration looks like this:

dial-peer voice 1 pots

description Peer for ISDN30

translation-profile incoming inbound-from-bt

service session

destination-pattern .T

progress_ind alert strip 8

no digit-strip

direct-inward-dial

port 1/0:15

forward-digits all

!

dial-peer voice 3 voip

tone ringback alert-no-PI

description Peer for bearer number

huntstop

service session

destination-pattern 01234567890

rtp payload-type nte-tone 102

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:1.2.3.4:5060

voice-class codec 1

dtmf-relay rtp-nte digit-drop

ip qos dscp cs3 signaling

clid network-provided

clid substitute name

!

dial-peer voice 65008 voip

tone ringback alert-no-PI

description Peer for main range

huntstop

service session

destination-pattern 01234567[1-8].

rtp payload-type nte-tone 102

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:1.2.3.4:5060

voice-class codec 1

dtmf-relay rtp-nte digit-drop

ip qos dscp cs3 signaling

clid network-provided

clid substitute name

Other relevant config:

voice rtp send-recv

!

voice service pots

rtcp keepalive

!

voice service voip

ip address trusted list

  ipv4 1.2.3.4

rtcp keepalive

dtmf-interworking standard

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

sip

  transport switch udp tcp

  no anat

  block 183 sdp present

!

voice class codec 1

codec preference 1 ilbc mode 30

codec preference 2 g729r8

codec preference 3 g711alaw

!

Now, when I place a call from ISDN through the gateway, the SDP it offers to my SBC is:

v=0

o=CiscoSystemsSIP-GW-UserAgent 7963 1810 IN IP4 6.7.8.9

s=SIP Call

c=IN IP4 6.7.8.9

t=0 0

m=audio 18710 RTP/AVP 116 18 8 101 13

c=IN IP4 6.7.8.9

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=30

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:13 CN/8000

This is great - both ends agree iLBC and the call comes up. However, if I make a call from the SIP side through to the ISDN, I offer this:

v=0

o=- 78207127932020 78207127932020 IN IP4 1.2.3.4

s=-

c=IN IP4 1.2.3.4

t=0 0

m=audio 24132 RTP/AVP 110 8 127

a=rtpmap:110 iLBC/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:110 mode=30

a=silenceSupp:on - - - -

a=ptime:20

It then gives me a 200 OK with this:

v=0

o=CiscoSystemsSIP-GW-UserAgent 9977 8349 IN IP4 6.7.8.9

s=SIP Call

c=IN IP4 6.7.8.9

t=0 0

m=audio 17650 RTP/AVP 8 127

c=IN IP4 6.7.8.9

a=rtpmap:8 PCMA/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-16

a=ptime:20

This means we use G.711 A-Law. If I remove this from the offer and just propose iLBC, the IAD rejects the call request.

Clearly it supports calls using G.711 over the ISDN and iLBC on the SIP side - how do I persuade it to allow calls initiated from the SIP side to work as well as calls initiated from the ISDN side?

Many thanks in advance for any suggestions!

Sean

1 ACCEPTED SOLUTION

Accepted Solutions
paolo bevilacqua
Hall of Fame Master

You should have "incoming called-number ." under voip DP.

View solution in original post

4 REPLIES 4
paolo bevilacqua
Hall of Fame Master

You should have "incoming called-number ." under voip DP.

OK, thanks - are you saying the incoming SIP call isn't matching one of the VoIP DPs so it's not picking up the codec config?

Paolo is correct; you're matching dial-peer 0 since there is nothing to match the incoming VOIP dial-peer on. In addition to the methods outlined in that document you can now also match based on the SIP URI.

Once you have added the command you can use the 'debug call active voice brief' command. The output will show the matched dialpeers. The pid will be the dial-peer tag number; if you see pid : 0 you're on dial-peer 0.

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

Thanks, that fixed it   I've now got a different issue with the IAD tearing the call down when it's put on hold on the SIP side with Q.850 release cause 172 and a '%HPI-3-FAILED_START: channel:1/0:15 DSP ID:0x2, failed mode 1 for service 27' log - we have a Music On Hold server that only does G.711, so it seems to be an issue switching codecs during the call. I'll create a separate thread for this though...

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