How to activate G722 audio codec

michael_zurich

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#1
Hi

How to activate G722 audio codec on elastix 1.2.4?

My elastix 1.2.4 shows with asterisk-cli "core show codecs" that G722 audio codec is there but when I try to make a G722 call there is no G722 in INVITE.

I use phoner which has G722 as first codec.

with Elastix I get no G722:
11:04:33,515: R: 192.168.1.96:5060 (UDP)
INVITE sip:706@192.168.1.67:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.96:5060;branch=z9hG4bK60f025e2;rport
From: "705" <sip:705@192.168.1.96>;tag=as0250a8ca
To: <sip:706@192.168.1.67:5060>
Contact: <sip:705@192.168.1.96>
Call-ID: 4c0ee98401284f452f079e88446f146e@192.168.1.96
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 20 Sep 2008 09:05:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 20547 20547 IN IP4 192.168.1.96
s=session
c=IN IP4 192.168.1.96
t=0 0
m=audio 16310 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

with LANCOM I have G722 working
11:08:44,968: R: 192.168.1.10:5060 (UDP)
INVITE sip:706@voiplan2.intern SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-bb3b3a90-8b04d1e2
From: "705"<sip:705@192.168.1.10;user=phone>;tag=-95254290-584128084
To: <sip:706@voiplan2.intern;user=phone>
Call-ID: 3259275999@00a05711fb86
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: LANCOM 1724 VoIP (Annex B) / 7.56.0046 / 20.08.2008
Server: lancom2
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Contact: <sip:705@192.168.1.10:5060>
Content-Type: application/sdp
Content-Length: 472

v=0
o=- 2358955599 2358955599 IN IP4 192.168.1.10
s=call
c=IN IP4 192.168.1.10
t=0 0
m=audio 52146 RTP/AVP 9 8 0 99 3 18 4 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-16
a=sendrecv
a=ptime:20

Michael
 

michael_zurich

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#2
I added in sip.conf in the section
[general]
allow=g722
but this does not help.

Any idea?
Michael
 

syadnom

Joined
Aug 4, 2009
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#3
asterisk 1.4 only does g.722 pass through. you need to patch asterisk to get g.722 translating.

you should be able to call between two g.722 compatible devices but you wont be able to call voicemail, or non g.722 devices, or place outgoing calls to PSTN or a voip provider that doesnt support g.722
 

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