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<font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.<br>
<br>
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.<br>
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.<br>
<br>
The SIP peers are both defined as :<br>
<br>
disallow=all<br>
allow=g726<br>
allow=alaw<br>
allow=g729<br>
allow=gsm<br>
<br>
<br>
<br>
This is the SIP trace :<br>
<br>
<br>
INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:20@192.168.1.150">sip:20@192.168.1.150</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9<br>
From: "User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.150"><sip:user@192.168.1.150></a>;tag=2383fb163ee6befa<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.102:5062;transport=udp"><sip:user@192.168.1.102:5062;transport=udp></a><br>
Supported: replaces, timer, path<br>
Proxy-Authorization: Digest username="user", realm="domain.be",
algorithm=MD5, uri=<a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150">"sip:20@192.168.1.150"</a>, nonce="1ae22736",
response="c90d0d9bf1f3c2bbc020651a5b67b608"<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
CSeq: 35396 INVITE<br>
<b>User-Agent: Grandstream GXP2010 1.2.1.4</b><br>
Max-Forwards: 70<br>
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE<br>
Content-Type: application/sdp<br>
Content-Length: 250<br>
<br>
v=0<br>
o=user 8000 8001 IN IP4 192.168.1.102<br>
s=SIP Call<br>
c=IN IP4 192.168.1.102<br>
t=0 0<br>
m=audio 10126 RTP/AVP 2 8 101<br>
a=sendrecv<br>
<b>a=rtpmap:2 G726-32/8000<br>
a=rtpmap:8 PCMA/8000</b><br>
a=ptime:20<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-11<br>
<br>
<-------------><br>
[Aug 2 13:56:57] --- (14 headers 12 lines) ---<br>
[Aug 2 13:56:57] Sending to 192.168.1.102 : 5062 (NAT)<br>
[Aug 2 13:56:57] Using INVITE request as basis request -
<a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
[Aug 2 13:56:57] Found user 'user'<br>
[Aug 2 13:56:57] Found RTP audio format 2<br>
[Aug 2 13:56:57] Found RTP audio format 8<br>
[Aug 2 13:56:57] Found RTP audio format 101<br>
[Aug 2 13:56:57] Found audio description format G726-32 for ID 2<br>
[Aug 2 13:56:57] Found audio description format PCMA for ID 8<br>
[Aug 2 13:56:57] Found audio description format telephone-event for ID
101<br>
<b>[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729),
peer - audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808
(alaw|g726)</b><br>
[Aug 2 13:56:57] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)<br>
[Aug 2 13:56:57] Peer audio RTP is at port 192.168.1.102:10126<br>
[Aug 2 13:56:57] Looking for 20 in from-STERKEN (domain 192.168.1.150)<br>
[Aug 2 13:56:57] list_route: hop:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.102:5062;transport=udp"><sip:user@192.168.1.102:5062;transport=udp></a><br>
[Aug 2 13:56:57] <br>
<--- Transmitting (NAT) to 192.168.1.102:5062 ---><br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102<br>
From: "User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.150"><sip:user@192.168.1.150></a>;tag=2383fb163ee6befa<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
CSeq: 35396 INVITE<br>
User-Agent: my-asterisk-server<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a><br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
[Aug 2 13:56:57] --- (11 headers 0 lines) ---<br>
[Aug 2 13:56:57] SIP Response message for INCOMING dialog NOTIFY
arrived<br>
[Aug 2 13:56:57] -- SIP/sterkendries2-00000054 is ringing<br>
[Aug 2 13:56:57] <br>
<--- Transmitting (NAT) to 192.168.1.102:5062 ---><br>
SIP/2.0 180 Ringing<br>
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102<br>
From: "User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.150"><sip:user@192.168.1.150></a>;tag=2383fb163ee6befa<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a>;tag=as655a8251<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
CSeq: 35396 INVITE<br>
<b>User-Agent: my-asterisk-server</b><br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a><br>
Content-Length: 0<br>
<br>
---<br>
[Aug 2 13:57:00] Extension Changed 20[105002-blf] new state InUse for
Notify User user <br>
[Aug 2 13:57:00] -- SIP/sterkendries2-00000054 answered
SIP/user-00000053<br>
[Aug 2 13:57:00] Audio is at 192.168.1.150 port 11500<br>
[Aug 2 13:57:00] Adding codec 0x8 (alaw) to SDP<br>
[Aug 2 13:57:00] Adding codec 0x800 (g726) to SDP<br>
[Aug 2 13:57:00] Adding non-codec 0x1 (telephone-event) to SDP<br>
[Aug 2 13:57:00] <br>
<--- Reliably Transmitting (NAT) to 192.168.1.102:5062 ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP
192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102<br>
From: "User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.150"><sip:user@192.168.1.150></a>;tag=2383fb163ee6befa<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a>;tag=as655a8251<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
CSeq: 35396 INVITE<br>
<b>User-Agent: my-asterisk-server</b><br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a><br>
Content-Type: application/sdp<br>
Content-Length: 267<br>
<br>
v=0<br>
o=root 1947 1947 IN IP4 192.168.1.150<br>
s=session<br>
c=IN IP4 192.168.1.150<br>
t=0 0<br>
m=audio 11500 RTP/AVP 8 2 101<br>
<b>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:2 G726-32/8000</b><br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
<-------------><br>
[Aug 2 13:57:00] --- (11 headers 0 lines) ---<br>
[Aug 2 13:57:00] SIP Response message for INCOMING dialog NOTIFY
arrived<br>
[Aug 2 13:57:00] <br>
<--- SIP read from 192.168.1.102:5062 ---><br>
ACK <a class="moz-txt-link-abbreviated" href="mailto:sip:20@192.168.1.150">sip:20@192.168.1.150</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK76a685e83ba8aef8<br>
From: "User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.150"><sip:user@192.168.1.150></a>;tag=2383fb163ee6befa<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150"><sip:20@192.168.1.150></a>;tag=as655a8251<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:user@192.168.1.102:5062;transport=udp"><sip:user@192.168.1.102:5062;transport=udp></a><br>
Supported: path<br>
Proxy-Authorization: Digest username="user", realm="domain.be",
algorithm=MD5, uri=<a class="moz-txt-link-rfc2396E" href="mailto:sip:20@192.168.1.150">"sip:20@192.168.1.150"</a>, nonce="1ae22736",
response="c90d0d9bf1f3c2bbc020651a5b67b608"<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:8910dbc6f2d5f71f@192.168.1.102">8910dbc6f2d5f71f@192.168.1.102</a><br>
CSeq: 35396 ACK<br>
<b>User-Agent: Grandstream GXP2010 1.2.1.4</b><br>
Max-Forwards: 70<br>
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE<br>
Content-Length: 0<br>
<br>
<br>
Question 1 :<br>
</font><font face="Helvetica, Arial, sans-serif"><b>[Aug 2 13:56:57]
Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer - audio=0x808
(alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)</b></font><br>
<font face="Helvetica, Arial, sans-serif"><br>
why is combined alaw|g726 and not g726|alaw (reverse) ??<br>
<br>
Question 2 :<br>
<br>
why do I see on my Grandstream phone that the codec being used is alaw
in stead of g726 ??<br>
<br>
Question 3 :<br>
<br>
How can I get g726 as first preferred codec ??<br>
<br>
<br>
<br>
<br>
Kind regards,<br>
<br>
Jonas.<br>
</font>
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