Voice cutoff in SIP trunk

Unanswered Question

We are experiencing voice cutoff with SIP trunk. the scenario is, when making outbound call from an IP phone to PSTN phone, CCM delivers through SIP trunk to GW, when far side answers the call, the first or two words are always missing.


However when we set up H323 trunk, and send the same call through H323 trunk to the same GW, no voice cutoff.


So there is definitely fine tuning in SIP, does anyone recommend something in SIP trunk (CCM) or sip (GW)? Thanks.


Wei

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gogasca Thu, 11/22/2007 - 19:50

Which version of CCM?

In 4.X we really dont have too much options for SIP Trunks. In 4.X I can think in Delay introduced by software MTP.

In 5.X and later you can try playing with SIP early media or other parameters under SIP Profile.

Please let us if you are using MTPs and topology.

A detailed CCM trace and debug ccsip messages

including sh run will be great.




It is CCM 6.0, no MTP is checked for this SIP trunk. we didn't configure anything in SIP prfile, all default, can you shred some light here?


Here are ccsip debug and sh run. If you could tell which CCM trace level, I could try to make it.


CCM: 142.125.119.43/44,

IP phone 142.125.119.114

GW: 142.125.119.94


Much appreciated.


Wei



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gogasca Sat, 11/24/2007 - 02:05

Hi Wei,

While checking the debugs, I found the following:


After Invite we see 183 session progress before 180.


We need to find a way to cut through audio earlier so that the Calling party can hear the ringing.


Nov 23 16:20:15.297: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x816D

Progress Ind i = 0x8088 - In-band info or appropriate now available


Nov 23 16:20:17.897: ISDN Se0/0/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x816D


With the 183 session progress - with SDP info, this means that RINGBACK SHOULD BE PLAYED INBAND.


If the gateway gets an ALERT/PROGRESS with a PI value of 1,2,8 it will always send 183


In CCM Admin 'Service parameters'

Scroll down to:Clusterwide Parameters (Device - SIP)


Select "True" in the SIP Rel1XX Enabled.


This forces the 1XX messages sent by the gateway to Callmanager to be Provisionally ACK'd.

The PRACK sent by Callmanager contains the SDP necessary to open the audio

path from the gateway to the IP phone.


The PRACk message would force an audio path to be temporarily opened till we get a second set of SDP from the connect message. If the SDP is the same the session stays up and if not a new once is instantaneously opened with the negotiated parameters in the SDP.


Let me know

Thanks

Hi, Gonzalo,


The fix you suggested actually works, I asked on site people to try, and no voice cut off any more. However the way I test remotely is still with voice cutoff.


The way I use to test remotely is CTIOS mobile agent, so call into CCM, and CCM initiates outbound call, then bridge these two calls together. We used to verifying that inbound call has no cutoff, so it is likely something happening during the bridging.


We do have TAC case open, thanks for all the help.

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