[asterisk-users] Codec g729 issues

[Saul Bejarano]

Hi Gustavo quick question:
As far as I can see the codec originated on the Huawei is g729
a=fmtp:18 annexb=yes

And on your asterisk
 > a=rtpmap:18 G729/8000
 >
 > a=fmtp:18 annexb=no

Check the options again for annexb
also make sure that ont he Huawei side you do not autonegotiate PCMU and 
PCMA leave only g729 option, also remember that Asterisk is a passthru 
free for g729 meaning if it does transcoding you need the license for 
asterisk if not the call passes through, I have noticed that I had to 
load all the IVR's in g729 in order for them to play correctly thought 
otherwise it is going to be a silent connection.

Kind regards,

Saul Bejarano

Gustavo A Gonzalez wrote:
> HI folks!  my topology is:
> 
>  
> 
>                softswitch (BROADSOFT) -- [sip trunk] -- Asterisk 
> 
>  
> 
> I need to connect phone calls using g729 codec. Debugging some calls we found that calls can’t connect because of codec incompatibility. Our Sip provider send us annexb=yes when a call is comming and our asterisk send annexb=no. I’m running asterisk 1.4.21.1. Output debug shows:
> 
>  
> 
> To: <sip:7002 at XXX.X.XXX.177;user=phone>
> 
> CSeq: 1 INVITE
> 
> Contact: <sip:000 at XXX.X.XXX.170:5060;user=phone;transport=udp>
> 
> Supported: 100rel
> 
> User-Agent: Huawei SoftX3000 V300R006
> 
> Max-Forwards: 69
> 
> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
> 
> Content-Length: 274
> 
> Content-Type: application/sdp
> 
>  
> 
> v=0
> 
> o=HuaweiSoftX3000 194579 194579 IN IP4 189.8.113.170
> 
> s=Sip Call
> 
> c=IN IP4 XXX.X.XXX.170
> 
> t=0 0
> 
> m=audio 49256 RTP/AVP 18 8 0 97
> 
> a=rtpmap:18 G729/8000
> 
> a=rtpmap:8 PCMA/8000
> 
> a=rtpmap:0 PCMU/8000
> 
> a=rtpmap:97 telephone-event/8000
> 
> a=fmtp:97 0-15
> 
> a=fmtp:18 annexb=yes
> 
>  
> 
> Via: SIP/2.0/UDP XXX.X.XXX.170:5060;branch=z9hG4bKo2echm202o5g2dc38701.1;received=XXX.X.XXX.170
> 
> From: <sip:000@ XXX.X.XXX.170;user=phone>;tag=25f94692
> 
> To: <sip:7002@ XXX.X.XXX.177;user=phone>;tag=as0de67360
> 
> Call-ID: 7b76a55fa7fb66a00d3cf9caca9d2d3b at 10.0.0.10
> 
> CSeq: 1 INVITE
> 
> User-Agent: Asterisk PBX
> 
> Supported: replaces
> 
> Contact: <sip:7002@ XXX.X.XXX.177>
> 
> Content-Type: application/sdp
> 
> Content-Length: 262
> 
>  
> 
> v=0
> 
> o=root 10183 10183 IN IP4 XXX.X.XXX.177
> 
> s=session
> 
> c=IN IP4 XXX.X.XXX.177
> 
> t=0 0
> 
> m=audio 10772 RTP/AVP 18 97
> 
> a=rtpmap:18 G729/8000
> 
> a=fmtp:18 annexb=no
> 
> a=rtpmap:97 telephone-event/8000
> 
> a=fmtp:97 0-16
> 
> a=silenceSupp:off - - - -
> 
> a=ptime:20
> 
> a=sendrecv
> 
>  
> 
> Thanks for any help!
> 
> *Gustavo A. González*
> Dto. de Infraestructura
> Despegar.com, Inc.
> ggonzalez at despegar.com <mailto:ggonzalez at despegar.com>
> 
>  
> 
> 
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