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Beginner

fax over sip codec problem

hi all,

i recently deployed UC560 on custormer site. for short period we used BRI for telephony. Now we are using sip trunk from local provider. Voice calls work fine but we are having trouble with fax. Provider said that we are using wrong codec(g729) on our side. In CCA we configured fxs port for fax. on sip trunk we configured all options that provider gave as. (codec G711..). When troubleshooting in CCA I noticed in PCM capture that when fax is ringing codec is G711, but when fax ansewers than codec shown is G729r8. Customer uses did translation and have sesicated number for fax. My question is Is there way to force (hardcode) UC to use only G711 codec for incoming and outgoing fax calls?

Thanx all in advance for all replies.

BR,

Goran                  

Everyone's tags (4)
1 ACCEPTED SOLUTION

Accepted Solutions
Rising star

Re: fax over sip codec problem

When a call doesn't match an existing dial-peer, the cisco uses an hidden dial-peer 0 that uses g729 codec.

How many dial-peer are configured?

Can you use the command "sh voice call status " during a call?

In this way we can see what dial-peer is used.

Regards.

View solution in original post

8 REPLIES 8
Rising star

fax over sip codec problem

You can do this in different ways.

- You can configure the fax-upspeeding adding these commands in your dial-peer:

      modem passthrough nse codec g711alaw

      fax rate disable

      fax protocol pass-through g711alaw

In this way, when UC DSP recognizes the FAX CED tone changes the codec to G.711.

- You can also force G.711 as only available codec replacing the "voice-class codec" with "codec g711a" in dial-peer.

- You can try to define specific dial-peers (incoming and outgoing) for fax number. Something like this:

dial-peer voice 15 voip

description FAX out

answer-address "YOUR FAX NUMBER"

modem passthrough nse codec g711alaw

session protocol sipv2

session target dns:"YOUR ITSP"

session transport udp

dtmf-relay rtp-nte

codec g711alaw

fax rate disable

fax protocol pass-through g711alaw

no vad

dial-peer voice 16 voip

description FAX in

incoming called-number "YOUR FAX NUMBER"

modem passthrough nse codec g711alaw

session protocol sipv2

session transport udp

dtmf-relay rtp-nte

codec g711alaw

fax rate disable

fax protocol pass-through g711alaw

no vad

Regards.

Beginner

fax over sip codec problem

Thanx for replay,

I tried everything but no luck I done some furder debuging and i think that provider is fooling us aronud. I think that its side is offering G729 codec and our side reply-s. Can you tell me how to now witch messages is from provider and whitch are reply

Rising star

fax over sip codec problem

Add the output of "debug ccsip messages".

Regards.

Beginner

fax over sip codec problem

Here is some from the start of the call.

006349: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:013436334@172.16.61.74:5060;user=phone SIP/2.0
Allow: UPDATE,REFER,INVITE,ACK,OPTIONS,REGISTER,SUBSCRIBE,MESSAGE,NOTIFY,BYE,CAN
CEL,PRACK
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
Contact: <172.16.1.20:5060>
Content-Type: application/sdp
CSeq: 621891236 INVITE
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
Max-Forwards: 30
Min-SE: 90
Session-Expires: 600
Supported: timer,100rel
To: <013436334>
User-Agent: Amis/HR (SBC)
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-2F0F-22B2254
Content-Length: 347

v=0
o=cp10 134069476827 134069476827 IN IP4 172.16.2.35
s=SIP Call
c=IN IP4 172.16.2.35
t=0 0
m=audio 31792 RTP/AVP 8 0 18 125 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

006350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-2F0F-22B2254
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
To: <013436334>;tag=34DECE0-1D46
Date: Tue, 26 Jun 2012 07:11:37 GMT
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
CSeq: 621891236 INVITE
Allow-Events: telephone-event
Min-SE:  1800
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


006351: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:013436334@172.16.61.74:5060;user=phone SIP/2.0
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
Contact: <172.16.1.20:5060>
CSeq: 621891236 ACK
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
Max-Forwards: 30
To: <013436334>;tag=34DECE0-1D46
User-Agent: Amis/HR (SBC)
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-2F0F-22B2254
Content-Length: 0


006352: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:013436334@172.16.61.74:5060;user=phone SIP/2.0
Allow: UPDATE,REFER,INVITE,ACK,OPTIONS,REGISTER,SUBSCRIBE,MESSAGE,NOTIFY,BYE,CAN
CEL,PRACK
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
Contact: <172.16.1.20:5060>
Content-Type: application/sdp
CSeq: 621891254 INVITE
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
Max-Forwards: 29
Min-SE: 1800
Session-Expires: 1800
Supported: timer,100rel
To: <013436334>
User-Agent: Amis/HR (SBC)
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-737B-22B2258
Content-Length: 347

v=0
o=cp10 134069476829 134069476829 IN IP4 172.16.2.35
s=SIP Call
c=IN IP4 172.16.2.35
t=0 0
m=audio 37208 RTP/AVP 8 0 18 125 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

006353: //6208/0100004183B1/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-737B-22B2258
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
To: <013436334>
Date: Tue, 26 Jun 2012 07:11:37 GMT
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
CSeq: 621891254 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


006354: //6208/0100004183B1/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-737B-22B2258
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
To: <013436334>;tag=34DED10-B86
Date: Tue, 26 Jun 2012 07:11:37 GMT
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
CSeq: 621891254 INVITE
Require: 100rel
RSeq: 776
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Contact: <013436334>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 5168 3647 IN IP4 172.16.61.74
s=SIP Call
t=0 0
m=audio 16684 RTP/AVP 18
c=IN IP4 172.16.61.74
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

006355: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:013436334@172.16.61.74:5060 SIP/2.0
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
CSeq: 621891255 PRACK
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
Max-Forwards: 29
RAck: 776 621891254 INVITE
To: <013436334>;tag=34DED10-B86
User-Agent: Amis/HR (SBC)
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-5FB-22B2259
Content-Length: 0


006356: //6208/0100004183B1/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-5FB-22B2259
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
To: <013436334>;tag=34DED10-B86
Date: Tue, 26 Jun 2012 07:11:37 GMT
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 621891255 PRACK
Content-Length: 0


006357: //6208/0100004183B1/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-737B-22B2258
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
To: <013436334>;tag=34DED10-B86
Date: Tue, 26 Jun 2012 07:11:37 GMT
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
CSeq: 621891254 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Contact: <013436334>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 5168 3647 IN IP4 172.16.61.74
s=SIP Call
t=0 0
m=audio 16684 RTP/AVP 18
c=IN IP4 172.16.61.74
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

006358: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:013436334@172.16.61.74:5060 SIP/2.0
Call-ID: 26762-TA-a524bec4-3ef4ac612@sip-priv.amis.hr
Contact: <172.16.1.20:5060>
CSeq: 621891254 ACK
From: "016637806" <>016637806@sip-priv.amis.hr;user=phone>;tag=26762-WF-a524b
ec5-7368e5290
Max-Forwards: 29
To: <013436334>;tag=34DED10-B86
User-Agent: Amis/HR (SBC)
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK-75EA-22B229D
Content-Length: 0


006359: Jun 26 09:11:44.696: %SEC-6-IPACCESSLOGP: list 107 denied tcp 188.129.18
5.218(9635) -> 188.129.6.126(445), 1 packet

...................

BR

Rising star

Re: fax over sip codec problem

This is the call flow:

- incoming INVITE from User-Agent: Amis/HR (SBC) with these codecs

a=rtpmap:8 PCMA/8000/1

a=rtpmap:0 PCMU/8000/1

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:125 CLEARMODE/8000/1

a=rtpmap:101 telephone-event/8000

and option Min-SE: 90

- outgoing response from your cisco

422 Session Timer too small - Min-SE:  1800

- incoming ReINVITE form User-Agent: Amis/HR (SBC) with these codecs

a=rtpmap:8 PCMA/8000/1

a=rtpmap:0 PCMU/8000/1

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:125 CLEARMODE/8000/1

a=rtpmap:101 telephone-event/8000

but new Min-SE: 1800

- outgoing final 200 OK response from your cisco with G729

m=audio 16684 RTP/AVP 18

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

Your provider supports G711. You can add in your configuration using voice-class codec and right priority.

Regards.

Beginner

Re: fax over sip codec problem

thank you very mutch, but I am realy stuck here in voice class codec i have only g711 i dond know wrom were uc pulls g729 to offer.

Rising star

Re: fax over sip codec problem

When a call doesn't match an existing dial-peer, the cisco uses an hidden dial-peer 0 that uses g729 codec.

How many dial-peer are configured?

Can you use the command "sh voice call status " during a call?

In this way we can see what dial-peer is used.

Regards.

View solution in original post

Beginner

Re: fax over sip codec problem

Thanx again very mutch for all your help.

Problem was in dial-peers. I managed to locate it by using command that you sugdested. Then I hardcoded g711alaw codec in configuration and matched configuration for incoming and outgoing dial-peer.

Best regards,

Goran