Problem CCME codec

divine007 · ‎06-25-2012

Hi,

I dont know if is possible to do global codec modification inside my CCME presently i believe the default is

g711ulaw which presently using. Can i do a global config to change this like to g729AB.

Thanks

Ayodeji Okanlawon · ‎06-25-2012

Yes you can.

Configure the codec you want under the phone..

eg

ephone 1

codec g729

or if you are using sip phone

voice register pool

codec g729

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divine007 · ‎06-25-2012

Sure, you correct. tell me if i have if i dont have all those codecs available on, do i have to upgrade my IOS??

Before i can have them?

Regards

Ayodeji Okanlawon · ‎06-25-2012

I dont think it is an IOS issue. What IP Phones are you using and what is the version of your CCME. Most cisco phones can support g729ab (g729 annex-b)

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divine007 · ‎06-24-2013

I have issues with my SIP trunk

thats from my ccme<==>sip server. here is the config

dial-peer voice 20 voip

description connect to Softswitch

destination-pattern 7890T

session protocol sipv2

session target ipv4:192.168.50.1

dtmf-relay rtp-nte

codec g711ulaw

When i call to from my cisco phone extension to my GSM number it rings twice after that call goes of

And when i debug using "debug ccsip media" this is what i have

Jun 24 15:17:51.755: //131/0ACD1405812F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 24 15:17:51.755: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17740 for stream 1

Jun 24 15:17:51.755: //131/0ACD1405812F/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101

Jun 24 15:17:51.755: //131/0ACD1405812F/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 24 15:17:51.755: //131/0ACD1405812F/SIP/Media/sipSPIProcessRtpSessions: No active streams.

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 131) to the VOIP RTP library

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 192.168.1.1, lport = 17740, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE

src_callid = 131, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY

media_ip_addr = 0.0.0.0, vrf tableid = 0

Jun 24 15:17:51.759: //131/0ACD1405812F/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one

Jun 24 15:17:59.483: //131/0ACD1405812F/SIP/Error/sipSPICheckSingleStreamCriteria: Codec negotiation failed on single stream

Jun 24 15:17:59.483: //131/0ACD1405812F/SIP/Error/ccsip_api_call_cut_progress: MediaNegotiation Failure - Send Cancel

Jun 24 15:17:59.483: //131/0ACD1405812F/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value

Jun 24 15:17:59.531: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIMatchRespToReqTran: Error in matching To header tags

Can someone advise me on this!!

Ayodeji Okanlawon · ‎06-24-2013

You need to send your config...and debug ccsip messages..

From the sample output it looks like a codec issue. But I cant tell why unless I see the debug and the sh run of your gateway

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divine007 · ‎06-24-2013

Here is the debug ccsip messages

Jun 25 05:49:19.793: //447/C8E71BA984DF/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 25 05:49:19.793: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 18856 for stream 1

Jun 25 05:49:19.797: //447/C8E71BA984DF/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101

Jun 25 05:49:19.797: //447/C8E71BA984DF/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 25 05:49:19.797: //447/C8E71BA984DF/SIP/Media/sipSPIProcessRtpSessions: No active streams.

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 447) to the VOIP RTP library

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 192.168.1.1, lport = 18856, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE

src_callid = 447, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY

media_ip_addr = 0.0.0.0, vrf tableid = 0

Jun 25 05:49:19.813: //447/C8E71BA984DF/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one

Jun 25 05:49:19.813: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:789075697183@192.168.50.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKE1AEE

Remote-Party-ID: "Raoul" <240>;party=calling;screen=no;privacy=off

From: "Raoul" <240>;tag=354C13C-C99

To: <789075697183>

Date: Tue, 25 Jun 2013 05:49:19 GMT

Call-ID: D05287A9-DC9111E2-84E48911-59004C21@192.168.1.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 3370589097-3700494818-2229242129-1493191713

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1372139359

Contact: <240>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 277

v=0

o=CiscoSystemsSIP-GW-UserAgent 5126 6537 IN IP4 192.168.1.1

s=SIP Call

c=IN IP4 192.168.1.1

t=0 0

m=audio 18856 RTP/AVP 0 101 19

c=IN IP4 4192.168.1.1

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

Ayodeji Okanlawon · ‎06-24-2013

This debug is incomplete. Pls send full debug..and your config. Attach them here

Sent from Cisco Technical Support Android App

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divine007 · ‎06-24-2013

Jun 25 06:20:41.864: //458/2AA5BFB684F3/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 25 06:20:41.864: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17522 for stream 1

Jun 25 06:20:41.868: //458/2AA5BFB684F3/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101

Jun 25 06:20:41.868: //458/2AA5BFB684F3/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 25 06:20:41.868: //458/2AA5BFB684F3/SIP/Media/sipSPIProcessRtpSessions: No active streams.

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 458) to the VOIP RTP library

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 192.168.1.1, lport = 17522, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE

src_callid = 458, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY

media_ip_addr = 0.0.0.0, vrf tableid = 0

Jun 25 06:20:41.884: //458/2AA5BFB684F3/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one

Jun 25 06:20:41.888: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:789075697183@192.168.50.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Remote-Party-ID: "Raoul" <240>;party=calling;screen=no;privacy=off

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>

Date: Tue, 25 Jun 2013 06:20:41 GMT

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 715505590-3700822498-2230552849-1493191713

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1372141241

Contact: <240>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 277

v=0

o=CiscoSystemsSIP-GW-UserAgent 2027 4440 IN IP4 192.168.1.1

s=SIP Call

c=IN IP4 192.168.1.1

t=0 0

m=audio 17522 RTP/AVP 0 101 19

c=IN IP4 192.168.1.1

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

Jun 25 06:20:41.924: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 INVITE

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER, PRACK

Content-Length: 0

Jun 25 06:20:46.992: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=JnT9nn7jSqLyqPDAMaD8A1msBK

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 INVITE

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER, PRACK

Content-Length: 0

Jun 25 06:20:46.996: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=GbQS3iZ3mXsJhLbXqp8BR8lCxO

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 INVITE

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER, PRACK

Content-Type: application/sdp

Content-Length: 222

v=0

o=- 1372141247 1372141247 IN IP4 192.168.50.1

s=anonymous

i= noinfo

c=IN IP4 192.168.40.1

t=0 0

m=audio 13550 RTP/AVP 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=sendrecv

a=silenceSupp:on - - - -

Jun 25 06:20:46.996: //458/2AA5BFB684F3/SIP/Error/sipSPICheckSingleStreamCriteria: Codec negotiation failed on single stream

Jun 25 06:20:46.996: //458/2AA5BFB684F3/SIP/Error/ccsip_api_call_cut_progress: MediaNegotiation Failure - Send Cancel

Jun 25 06:20:47.000: //458/2AA5BFB684F3/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value

Jun 25 06:20:47.004: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:789075697183@192.168.50.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>

Date: Tue, 25 Jun 2013 06:20:41 GMT

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1372141247

Reason: Q.850;cause=65

Content-Length: 0

Jun 25 06:20:47.012: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:789075697183@192.168.50.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>

Date: Tue, 25 Jun 2013 06:20:41 GMT

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1372141247

Reason: Q.850;cause=65

Content-Length: 0

Jun 25 06:20:47.020: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=1YLtlQVEAfAEtK5daELiIAqR5c

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 CANCEL

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Content-Length: 0

Jun 25 06:20:47.024: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=JnT9nn7jSqLyqPDAMaD8A1msBK

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 INVITE

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER, PRACK

Content-Length: 0

Jun 25 06:20:47.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:789075697183@192.168.50.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=JnT9nn7jSqLyqPDAMaD8A1msBK

Date: Tue, 25 Jun 2013 06:20:41 GMT

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Jun 25 06:20:47.048: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIMatchRespToReqTran: Error in matching To header tags

Jun 25 06:20:47.048: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=1YLtlQVEAfAEtK5daELiIAqR5c

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 CANCEL

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Content-Length:

Ayodeji Okanlawon · ‎06-25-2013

This problem is from your service provider...Your provider is not selecting/sending any codec in the answer to your INVITE. The INVITE you sent has G711ulaw, but the answer from your provider doesnt have any codec. Please contact them and ask them why they are not offering you any codec...You can show the logs.

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

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divine007 · ‎06-25-2013

wow

will get back to you

divine007 · ‎06-28-2013

Please can i do know the "line" which shows where the error is coming from?

Ayodeji Okanlawon · ‎06-28-2013

Here is the error..Thi is the 183 Session progress received from your ITSP..It has no codec in it.

Jun 25 06:20:46.996: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKF1C5F

Record-Route: <192.168.50.1:5060>

From: "Raoul" <240>;tag=3717908-6A4

To: <789075697183>;tag=GbQS3iZ3mXsJhLbXqp8BR8lCxO

Call-ID: 321FD1D7-DC9611E2-84F88911-59004C21@192.168.1.1

CSeq: 101 INVITE

Contact: NSProxy <192.168.50.1:5060>

User-Agent: N-SOFT-UA-5.25.4 -RIN1-2939- www.n-soft.com

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER, PRACK

Content-Type: application/sdp

Content-Length: 222

v=0

o=- 1372141247 1372141247 IN IP4 192.168.50.1

s=anonymous

i= noinfo

c=IN IP4 192.168.40.1

t=0 0

m=audio 13550 RTP/AVP 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=sendrecv

a=silenceSupp:on - -

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

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