[asterisk-users] G.729 codec in pass-thru mode

[Matthew J. Roth]

Kamlesh Kumar wrote:
> 
> SIP.conf
> [100]
> username=100
> secret=password
> type=friend
> host=dynamic
> nat=yes
> canreinvite=no
> insecure=port
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> context=asterisk
> qualify=no

Is there also an "allow=g729" line in sip.conf for the ITSP's SIP peer?

> SIP Trace: 
> 201.xxx.xxx.xxx = SIP Softphone which originates the call 
> xxx.xxx.xxx.xxx = Asterisk server 
> yyy.yyy.yyy.yyy = ITSP 
> 
> ...
> 
> <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060
> From: "100" <sip:100 at xxx.xxx.xxx.xxx>;tag=as643c20b1
> To: <sip:12127773456 at yyy.yyy.yyy.yyy>;tag=gK029aaa8c
> Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20 at xxx.xxx.xxx.xxx
> CSeq: 102 INVITE
> Contact: <sip:12127773456 at yyy.yyy.yyy.yyy:5060>
> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
> Content-Length:  234
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
> v=0
> o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy
> s=SIP Media Capabilities
> c=IN IP4 zzz.zzz.zzz.zzz
> t=0 0
> m=audio 21996 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=maxptime:20
> <------------->
> [Jun  3 13:11:31] --- (11 headers 11 lines) ---
> [Jun  3 13:11:31] Found RTP audio format 0
> [Jun  3 13:11:31] Found RTP audio format 101
> [Jun  3 13:11:31] Found audio description format PCMU for ID 0
> [Jun  3 13:11:31] Found audio description format telephone-event for ID 101
> [Jun  3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
> [Jun  3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
> [Jun  3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996
> [Jun  3 13:11:31]     -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress passing it to SIP/100-000034d8
> [Jun  3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042
> [Jun  3 13:11:31] Adding codec 0x4 (ulaw) to SDP
> [Jun  3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP

This response from the ITSP says that only u-law may be used for the call.
Please contact the ITSP and confirm that they actually support the G.729 codec.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer