Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.
The SIP peers are both defined as :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
This is the SIP trace :
INVITE sip:2...@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9
From: "User" <sip:u...@192.168.1.150>;tag=2383fb163ee6befa
To: <sip:2...@192.168.1.150>
Contact: <sip:u...@192.168.1.102:5062;transport=udp>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="user", realm="domain.be",
algorithm=MD5, uri="sip:2...@192.168.1.150", nonce="1ae22736",
response="c90d0d9bf1f3c2bbc020651a5b67b608"
Call-ID: 8910dbc6f2d5f...@192.168.1.102
CSeq: 35396 INVITE
*User-Agent: Grandstream GXP2010 1.2.1.4*
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 250
v=0
o=user 8000 8001 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 10126 RTP/AVP 2 8 101
a=sendrecv
*a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000*
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
[Aug 2 13:56:57] --- (14 headers 12 lines) ---
[Aug 2 13:56:57] Sending to 192.168.1.102 : 5062 (NAT)
[Aug 2 13:56:57] Using INVITE request as basis request -
8910dbc6f2d5f...@192.168.1.102
[Aug 2 13:56:57] Found user 'user'
[Aug 2 13:56:57] Found RTP audio format 2
[Aug 2 13:56:57] Found RTP audio format 8
[Aug 2 13:56:57] Found RTP audio format 101
[Aug 2 13:56:57] Found audio description format G726-32 for ID 2
[Aug 2 13:56:57] Found audio description format PCMA for ID 8
[Aug 2 13:56:57] Found audio description format telephone-event for ID 101
*[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)*
[Aug 2 13:56:57] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Aug 2 13:56:57] Peer audio RTP is at port 192.168.1.102:10126
[Aug 2 13:56:57] Looking for 20 in from-STERKEN (domain 192.168.1.150)
[Aug 2 13:56:57] list_route: hop:
<sip:u...@192.168.1.102:5062;transport=udp>
[Aug 2 13:56:57]
<--- Transmitting (NAT) to 192.168.1.102:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102
From: "User" <sip:u...@192.168.1.150>;tag=2383fb163ee6befa
To: <sip:2...@192.168.1.150>
Call-ID: 8910dbc6f2d5f...@192.168.1.102
CSeq: 35396 INVITE
User-Agent: my-asterisk-server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:2...@192.168.1.150>
Content-Length: 0
<------------->
[Aug 2 13:56:57] --- (11 headers 0 lines) ---
[Aug 2 13:56:57] SIP Response message for INCOMING dialog NOTIFY arrived
[Aug 2 13:56:57] -- SIP/sterkendries2-00000054 is ringing
[Aug 2 13:56:57]
<--- Transmitting (NAT) to 192.168.1.102:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102
From: "User" <sip:u...@192.168.1.150>;tag=2383fb163ee6befa
To: <sip:2...@192.168.1.150>;tag=as655a8251
Call-ID: 8910dbc6f2d5f...@192.168.1.102
CSeq: 35396 INVITE
*User-Agent: my-asterisk-server*
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:2...@192.168.1.150>
Content-Length: 0
---
[Aug 2 13:57:00] Extension Changed 20[105002-blf] new state InUse for
Notify User user
[Aug 2 13:57:00] -- SIP/sterkendries2-00000054 answered
SIP/user-00000053
[Aug 2 13:57:00] Audio is at 192.168.1.150 port 11500
[Aug 2 13:57:00] Adding codec 0x8 (alaw) to SDP
[Aug 2 13:57:00] Adding codec 0x800 (g726) to SDP
[Aug 2 13:57:00] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 2 13:57:00]
<--- Reliably Transmitting (NAT) to 192.168.1.102:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102
From: "User" <sip:u...@192.168.1.150>;tag=2383fb163ee6befa
To: <sip:2...@192.168.1.150>;tag=as655a8251
Call-ID: 8910dbc6f2d5f...@192.168.1.102
CSeq: 35396 INVITE
*User-Agent: my-asterisk-server*
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:2...@192.168.1.150>
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 1947 1947 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 11500 RTP/AVP 8 2 101
*a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000*
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Aug 2 13:57:00] --- (11 headers 0 lines) ---
[Aug 2 13:57:00] SIP Response message for INCOMING dialog NOTIFY arrived
[Aug 2 13:57:00]
<--- SIP read from 192.168.1.102:5062 --->
ACK sip:2...@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK76a685e83ba8aef8
From: "User" <sip:u...@192.168.1.150>;tag=2383fb163ee6befa
To: <sip:2...@192.168.1.150>;tag=as655a8251
Contact: <sip:u...@192.168.1.102:5062;transport=udp>
Supported: path
Proxy-Authorization: Digest username="user", realm="domain.be",
algorithm=MD5, uri="sip:2...@192.168.1.150", nonce="1ae22736",
response="c90d0d9bf1f3c2bbc020651a5b67b608"
Call-ID: 8910dbc6f2d5f...@192.168.1.102
CSeq: 35396 ACK
*User-Agent: Grandstream GXP2010 1.2.1.4*
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
Question 1 :
*[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)*
why is combined alaw|g726 and not g726|alaw (reverse) ??
Question 2 :
why do I see on my Grandstream phone that the codec being used is alaw
in stead of g726 ??
Question 3 :
How can I get g726 as first preferred codec ??
Kind regards,
Jonas.
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