No ring tone while testing SIP to SIP gateway

[meandaneth]

Dear All,

I have another issue now when calling from SIP to SIP, I have test call from SIP phone to destination number ar USA the call seem reach terminate gateway but not. There is no any ring back tone on my Sip phone. i have capture by wireshare and seem no rtp packet send back from my originate gateway to Sip server ip.

Anyway if i test from the same sip phone using H323 the call were fine with ringtone and connected properly.

Here is the scenario detail:

1. SIP Phone  ----- >SIP Server ----- > Originate GW(AS5400) ---- > Terminate GW ( ACME )   ========>>>>>> No Ring tone and not connect.

<---------------------------------------------  SIP Protocol ------------------------------------------------------------>

2. PSTN  ----- > Originate GW(AS5400)  ---- > Terminate GW ( ACME )   ============>>>>>  Call successfull properly.

<------------------TDM Link ------------> <-----------SIP Protocol--------------------->

3. SIP Phone  ----- >SIP Server ----- > Originate GW(AS5400) ---- > Terminate GW ( Nextone )   ============>>>>> Call successfull properly.

<------------------SIP Protocol ------------>  <-------------------------------H323------------------------------------->

Below is our dialpeer configuration in AS5400.

dial-peer voice 30833 voip
description To Aincent
translation-profile outgoing 318855
huntstop
preference 1
destination-pattern 31885519
voice-class codec 225
voice-class h323 1
session protocol sipv2
session target ipv4:202.83.198.7
dtmf-relay h245-alphanumeric

Please kinldy help me what can be resolve with this issue?

Best Regards,

Daneth

[Daniele Giordano]

Try to configure your AS5400 like an IP2IP signaling gateway:

voice service voip
allow-connections sip to sip

In this case the SIP flow is not direct from SIP Phone to Terminate GW (ACME) but is terminated and regenerated from AS5400.

1. SIP Phone  ----- >SIP Server ----- > Originate GW(AS5400)  ---- > Terminate GW ( ACME )   ========>

<-----------------------  SIP Protocol ---------------------------><-----------------------  SIP Protocol --------------------------->

If you want manage in this way RTP also you can use commands:

media flow-through

default is media flow-around

Best regards.

[meandaneth]

Dear Giordano,

I tried to configure this command in dial peer but it won't accept it and return invalid input as below:

gw-01-pnh-noc(config)#dial-peer voice 30833 voip
gw-01-pnh-noc(config-dial-peer)#media flow-through
                                 ^
% Invalid input detected at '^' marker.

My 5400 Software (C5400-IS-M), Version 12.4(11)T4

Best Regards,

Daneth

[Daniele Giordano]

Can you insert IP ADDRESSES present on attached screen shot in your scheme?
Can you post the pcap file?

We must study SIP contact header and SDP header to find the problem.

Thanks.

[meandaneth]

Dear Giordano,

Thanks for your help.Please kindly check my pcap file in attache below:

and  wireshark filter was:

sip && ((ip.src>=119.82.250.19 || ip.src>=119.82.248.15 && ip.dst==202.83.198.7 ) || (ip.src==202.83.198.7 && ip.dst>=119.82.250.19 || ip.dst>=119.82.248.15 ))

Best regards,

Daneth

[Daniele Giordano]

I've tried to imagine your network.
Is the scheme attached right?

I've tried to imagine your network.
Is the scheme attached right?
If yes, the SIP Signalling between AS5400 and Nextone is working correctly.
The problem may be in communication between AS5400 and SipPhone.
I say this because the AS5400 receive the "180 ringing" by the Nextone but doesn't forward it to the phone.

Another comment.
I think that your AS5400 is configured like an IP2IP gateway. Right? The sip "call-id"of the received INVITE is different than sent.
And a question.
What is the device with IP Address 119.82.248.17? Is the SIPPhone? This IP address is present in the SDP part of first SIP message.

Are you sure of IP reachability? From AS5400 can you ping the SIP Server and SIP Phone address?

Try  "debug ccsip message" on AS5400 and post the output.

Best regards.

[meandaneth]

Dear Giordano,

Yes we are configure IP2IP and below is the log while capture ccsip.

You can see calling number 77756127 and called number 33614260122 in attache.

it look like gateway try to invite many time but not success.

Best Regards,

Danet

[Daniele Giordano · Accepted solution]

dial-peer voice 30833 voip

description To Aincent

translation-profile outgoing 318855

huntstop

preference 1

destination-pattern 31885519

voice-class codec 225

voice-class h323 1 

session protocol sipv2

session target ipv4:202.83.198.7

dtmf-relay h245-alphanumeric   <== in a SIP dial-peer you can use dtmf-relay rtp-nte, this enable RFC2833 DTMF signalling

[Senthil Kumar Sankar]

Hi

When i looked into the trace found that AS5400 recieves 180 ringing with the header Require : 100rel from 202.83.198.7,

Does the AS5400 shud respond with PRACK.??

Please correct me If I am wrong.

[Daniele Giordano]

In the INVITE message sent from AS5400 there is this string:
Supported: 100rel,timer,resource-priority,replaces
So the AS5400 says "I support 100rel and if the other end require it I send a PRACK".
But this does not occur.

The 180 ringing from NEXTONE includes SDP.
This means that no ring back tone but early media message must be provided from terminal gateway.

Try to configure the AS5400 to ignore this:

AS5400(config)# sip-ua
AS5400(config-sip-ua)# disable-early-media 180

This command provides the ability to enable or disable early media cut-through on Cisco IOS gateways for Session Initiation Protocol (SIP) 180 responses with SDP. Use the disable-early-media 180 command to configure the gateway to ignore the SDP message and provide local ringback. To restore the default treatment, early media cut-through, use the no disable-early-media 180 command.

[meandaneth]

Dear Giordano,

I have tested again and disable vad in my voip phone and the call can go thru. I am not clear of this process but i try to disable vad in dial-peer both in-out going dial-peer the call still not go thru,unless disable vad in voipphone.

what could be the reason and please advice if the default setting of other voip phone is enable vad.

Best Regards,

Daneth

[Daniele Giordano]

mmm, not easy

I think that "vad" affects codec negotiation.
You use G729 codec. In same cases "vad" enables or disables the annex b variant of this codec. In other cases "vad" affects also packetization time.

This is your SIP/SDP INVITE:

a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
a=ptime:30

What happens when you disable "vad"?

If "vad" is disable it must be announced because no mentioning annexb is interpreted as annexb=yes.

The annex b can also varies the ptime.
In same cases if ptime doesn't match can be cause of problems.
In the past I had problems with Nextone SBC. It blocked calls because my ptime was different by expected.

Your solution is probably correct. The "vad" must be disabled or enabled on each device to avoid problems of interworking.

Best regards.

[meandaneth]

Dear Giordano,

This is still problem with vad enable, the call still not go thru while enable vad in ipphone. However this solution in not good while there are different other customer who may enable it by default ant it will affect to the performace.

I have captured both file with diable and enable vad for comparison but not much understand what was wrong. please kindly advice on this case while we need call success even enable or disable vad in ipphone.

Thanks and Regards,

Daneth

[Daniele Giordano · Accepted solution]

This is the message sent by your phone:

INVITE sip:31785510356011@119.82.250.12:5060 SIP/2.0
Via: SIP/2.0/UDP 119.82.248.16:5060;branch=z9hG4bK1292407982905,SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bK8583c2939c
From: "false" <77756127>;tag=24fbd484
To: <31785510356011>
Call-ID: 333a438e362939e84c98ca1f42fde082@192.168.1.110
CSeq: 2 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Allow-Events: talk,hold,conference
Expires: 300
Contact: "false" <77756127>
Remote-Party-ID: "false" <77756127>;screen=yes;party=calling;privacy=off
P-Asserted-Identity: "false" <77756127>
Privacy: none
Content-Type: application/sdp
Content-Length: 255

v=0
o=CMI-SIPUA 48246 0 IN IP4 119.82.248.17
s=SIP CALL
c=IN IP4 119.82.248.17
t=0 0
m=audio 22038 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=yes  ===> VAD enable and so enable G.729b variant
a=ptime:20
a=rtcp:60001
a=sendrecv

This is the message sent by your cisco AS5400:

INVITE sip:85510356011@203.118.242.45:5060 SIP/2.0
Via: SIP/2.0/UDP 119.82.250.12:5060;branch=z9hG4bK3D2B8
Remote-Party-ID: "false" <77756127>;party=calling;screen=yes;privacy=off
From: "false" <77756127>;tag=18EE7968-1161
To: <85510356011>
Date: Wed, 15 Dec 2010 10:13:02 GMT
Call-ID: BC5F9F0E-76A11E0-8263FAAF-726B99FA@119.82.250.12
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 3160263318-124391904-2187328175-1919654394
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1292407982
Contact: <77756127>
Call-Info: <119.82.250.12:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 346

v=0
o=CiscoSystemsSIP-GW-UserAgent 653 5246 IN IP4 119.82.250.12
s=SIP Call
c=IN IP4 119.82.250.12
t=0 0
m=audio 23152 RTP/AVP 18 100 101 19
c=IN IP4 119.82.250.12
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes  ===> preserve VAD
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

This is the message received by your AS5400:

SIP/2.0 180 Ringing
From: "false"<77756127>;tag=18EE7968-1161
To: <85510356011>;tag=50da1260
Call-ID: BC5F9F0E-76A11E0-8263FAAF-726B99FA@119.82.250.12
CSeq: 101 INVITE
Record-Route: <203.118.242.45:5060>
Server: vocl-essentra-ex/8.0F2 (19020.24)
Supported: 100rel
Contact: <010356011>
Via: SIP/2.0/UDP 119.82.250.12:5060;branch=z9hG4bK3D2B8
Require: 100rel
RSeq: 612
Content-Type: application/sdp
Content-Length: 181

v=0
o=Essentra-Relay 3888415630 1 IN IP4 203.118.242.45
s=-
c=IN IP4 203.118.242.45
t=0 0
m=audio 37310 RTP/AVP 18
a=fmtp:18 annexb=no  ===> no VAD
a=ptime:20
a=silenceSupp:off - - - -

And these are internal operations performed by cisco AS5400:

sipSPISelectCodecVersion: Codec (g729r8) is not in preferred list
sipSPIDoAudioNegotiation: No matching voice codec found for m-line 1
sipSPIStreamTypeAndDtmfRelay: No voice codec and no dtmf-relay match
sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
sipSPIDoAudioNegotiation: Media negotiation failed for m-line 1
ccsip_api_call_alert: MediaNegotiation Failure - Send Cancel
ccsip_api_call_alert returned: SIP_INTERNAL_ERR
sipSPISendCancel: Sending CANCEL to the transport layer
sipSPITransportSendMessage: Proceedable for sending msg immediately

and so your AS5400 sent:

CANCEL sip:85510356011@203.118.242.45:5060 SIP/2.0
Via: SIP/2.0/UDP 119.82.250.12:5060;branch=z9hG4bK3D2B8
From: "false" <77756127>;tag=18EE7968-1161
To: <85510356011>
Date: Wed, 15 Dec 2010 10:13:02 GMT
Call-ID: BC5F9F0E-76A11E0-8263FAAF-726B99FA@119.82.250.12
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1292407986
Reason: Q.850;cause=65
Content-Length: 0

In your dial-peer you configured a voice-class codec 225.
Try to add G.729 variant in your voice-class codec:

voice class codec 225
codec preference 1 g729r8
codec preference 2 g729br8

and other codec used

When you apply g729br8 this warning message appear:

WARNING: This codec has built-in vad that cannot be disabled.
If this codec is negotiated the vad configuration will be disregarded.

Best regard.

[meandaneth]

Dear Giordano,

Thanks for your good explaination.

Best Regards,

Danet