SIP Dialer codec

[Yankovskyi]

Hello,

There is one question to community:

Where we can setup preferable codec in Cisco SIP dialer?

It is needed to change g711ulaw which is default SIP Dialer codec to g711alaw. 

Thanks in advance,

Regards,

Alex

[Chintan Gajjar]

i guess this can be done at Voice Gateway Dial-peer level, by putting it in inbound and Outbound dial-peers.

regards

Chintan

[Yankovskyi]

Hi Chintain,

thanks for response.

Yes it's right we can create settings on dialpeers, but I don't have any dsp resources and PSTN provider accept only g711alaw. So I am try to find way how to change default SIP dialer codec from g711ulaw to g711alaw.

regards,

Alex

[Chintan Gajjar]

Alex, 

As per my Understanding, there is no Default Codec with Sip Dialer.

This configurations are all dependent on Voice Gateway, If you configure Voice gateway to use G711alaw(Based on Dial-peer) the dialer will use that codec.

there is a default codec for Cisco VG, and that is g 729. but you can avoid using that with your setting.

i think you are concerned about codec conversion, so if all will be in single codec you will not need any Transcoding, But definitely you will need DSP for Converting Voip traffic into TDM and Vice Versa. 

regards

Chintan

~please rate all helpful posts

[Yankovskyi]

Chintan,

I see that INVITE from dialer comes with one codec g711ulaw. Please see below:

INVITE sip:0675498693@10.13.81.27 SIP/2.0
Via: SIP/2.0/UDP 10.13.81.3:58800;branch=z9hG4bK-d8754z-57145a04a377e42c-1---d8754z-;rport
Max-Forwards: 70
Require: 100rel
Contact: <sip:8999909@10.13.81.3:58800>
To: <sip:0675498693@10.13.81.27>
From: <sip:8999909@10.13.81.3>;tag=23395838
Call-ID: eb13700c-00095a52-50032441-92184f3b
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS
 --More--         
Content-Type: Multipart/mixed;boundary=uniqueBoundary
Supported: timer, resource-priority, replaces
User-Agent: Cisco-SIPDialer/UCCE8.0
Content-Length: 608
Remote-Party-ID: <sip:@10.13.81.27>;party=calling;screen=no;privacy=off

--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=CiscoSystemsSIP-GW-UserAgent 2884 2524 IN IP4 172.19.155.41
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 19994 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--uniqueBoundary
 --More--         
Content-Type: application/x-cisco-cpa
Content-Disposition: signal;handling=optional

Events=FT,Asm,AsmT,Sit,Piano
CPAMinSilencePeriod=608
CPAAnalysisPeriod=2500
CPAMaxTimeAnalysis=3000
CPAMaxTermToneAnalysis=30000
CPAMinValidSpeechTime=112

Regards,

Alex