Voxeo SIP Phone Calls dropping thru Siemens Hicom300 during silence

sdoherty · ‎02-05-2008

HEllo

We have a developer testing a new Voxeo XML app placing a call via SIP thru the router via ISDN to the HICOM and then to the PSTN. When there is silence on the line the call drops. Any ideas? thank you

pcameron · ‎02-05-2008

Since you have given us absolutely no clue about the gateway platform, IOS versions, configs and detailed prpblem symptoms etc ... it's going to be hard to give a definitive answer, but assuming your application is working through an IOS gateway, the most common reason why calls drop after periods of silence is the recieve RTP or recieve RTCP timer expires after an extended period of silence, which means no RTP packets come into the gateway. As a result the router assumes the call has dropped and clears the call. You can extend these timers here -

CME-Switch#conf t

Enter configuration commands, one per line. End with CNTL/Z.

CME-Switch(config)#gateway

CME-Switch(config-gateway)#

CME-Switch(config-gateway)#media-inactivity-criteria ?

all Both RTP and RTCP for silence detection

rtcp Use RTCP for silence detection

rtp Use RTP stream for silence detection, this is default

CME-Switch(config-gateway)#timer ?

media-inactive (dsp based) configure media inactivity timer- available for

h323/sip only at this time

receive-rtcp (non-dsp based) configure media inactivity timer- available

for h323/sip only at this time

receive-rtp configure absolute RTP receive timer in seconds

CME-Switch(config-gateway)#

Try setting these timers to see if they make a difference.

sdoherty · ‎02-06-2008

Hello

thanks pcameron - I tried all the settings and still the calls timeout. We are using an IOS based router and today we ran the T1 directly to the router and not the PBX and same issue- calls drop after 10-20 secs of silence. Here is a debug ccsip;

*Feb 6 19:09:19.030: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 156.132.20.200:5060;branch=z9hG4bKa3c18a18f7998c33dd65a5442705f

38a973b525628eafd8a8f975db41ac4a6ff5060,SIP/2.0/UDP 156.132.219.235:5070

From: ;tag=0-13ce-47aa02e6-6289df3-2ea6

To: <9120232XXXXXX>;tag=3EEA94-EEC

Date: Wed, 06 Feb 2008 19:09:14 GMT

Call-ID: 0-13ce-47aa02e6-6289df3-bb3-1b0b608

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF

Y, INFO, REGISTER

Allow-Events: telephone-event

Contact: <91202327XXXX>

Record-Route: <156.132.20.200:5060>

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 253

v=0

o=CiscoSystemsSIP-GW-UserAgent 7318 5046 IN IP4 156.132.34.202

s=SIP Call

c=IN IP4 156.132.34.202

t=0 0

m=audio 17336 RTP/AVP 0 101

c=IN IP4 156.132.34.202

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

pcameron · ‎02-06-2008

Paste a copy of the SH RUN and SH VER from the router, then enabled 'debug ccsip messages ' and 'debug isdn q931'. Make a single test call (too hard to follow if multiple calls happening) that shows the problem, and paste the logs here.

sdoherty · ‎02-07-2008

I am replying again as the first did not post. Here is the information you requested - thanks again. BTW our helpdesk is using the same router/pbx on a SIP/VOIP system with no problems...