Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: wyeee on June 16, 2011, 11:09:18 AM
-
I am using some softphones connected to Asterisk to test G.711 codec for Mitel 3300.
* INVITE from Asterisk to Mitel 3300 has rtpmap for 0 (uLaw), 8 (aLaw), 18 (G.729), and 101 (telephone-event).
* 200 OK from Mitel to Asterisk has only rtpmap of 101 (telephone-event)
(If I used a generic sip phone for Mitel, I can also see INVITE from 3300 to sip phone has no codec and
200 OK from sip phone has 0,8,18,101 in codec list)
The voice path is OK. I verified that between my softphone and Asterisk uLaw is used.
Two questions:
1. Why 0 (uLaw) is not include in the 200 OK from 3300 to Asterisk? I thought 3300 support both G.711 and G.729 (just learned that need to change zone info for G.729 to work).
2. What is the actual codec being used between 3300 and Asterisk (SDP contains no codec but telephone-event)? Uncompressed/raw voice? uLaw? (I know Asterisk is capable of transcoding if there is a mismatch)
Thanks a lot
-
I would have to setup a new box and get a capture but all the past captures I have looked at and seen the Mitel supports 711 and 729. As you said in order to use 729 you need to be in a different zone.
-
Are you trying to connect to the Mitel as an extension or a SIP trunk?
Ralph
-
Thanks a lot guys.
I am doing SIP trunking test between Mitel 3300 and Asterisk.
We have our own PBX in between (so there is a SIP trunk between Asterisk and our PBX and there is another SIP trunk between our PBX and Mitel 3300). Our PBX is not anchoring the call (it just routes it, SDP is not changed).
And here is the log (softphone 2408150 off Asterisk calls Mitel phone running Minet, 2258100. Logs are captured from our PBX):
Incoming Ingress Request
INVITE sip:2258100@192.168.80.146:5080 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.80.120:5060;branch=z9hG4bK53c653ea;rport=5060^M
From: "3CX admin" <sip:2408150@192.168.80.120>;tag=as2d832122^M
To: <sip:2258100@192.168.80.146:5080>^M
Contact: <sip:2408150@192.168.80.120>^M
Call-ID: 499274757b8ca33243049f5379eb9d70@192.168.80.120^M
CSeq: 102 INVITE^M
User-Agent: Asterisk PBX^M
Max-Forwards: 70^M
Date: Thu, 16 Jun 2011 18:19:37 GMT^M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY^M
Content-Type: application/sdp^M
Content-Length: 289^M
^M
v=0^M
o=root 1888 1888 IN IP4 192.168.80.120^M
s=session^M
c=IN IP4 192.168.80.120^M
t=0 0^M
m=audio 12592 RTP/AVP 0 8 18 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:18 G729/8000^M
a=fmtp:18 annexb=no^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=silenceSupp:off - - - -^M
Outgoing Egress Request
INVITE sip:2258100@mitel1.tango-networks.com:5060;transport=udp;maddr=192.168.80.143 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.80.146:5080;branch=z9hG4bK16409b6ecd0e22515d2d247448c33bf7;rport^M
From: <sip:2408150@system;transport=udp>;tag=TiBriA^M
To: <sip:2258100@mitel1.tango-networks.com;transport=udp>^M
Contact: <sip:192.168.80.146:5080;transport=udp;lr>^M
Call-ID: cf2412c3-f200-4379-a38a-7947b2fc44f6^M
CSeq: 100 INVITE^M
Max-Forwards: 69^M
Date: Thu, 16 Jun 2011 18:19:37 GMT^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
Content-Type: application/SDP^M
Content-Length: 289^M
P-Asserted-Identity: <sip:2408150@system;transport=udp>^M
Remote-Party-ID: <sip:2408150@system;transport=udp>;party=calling^M
^M
v=0^M
o=root 1888 1888 IN IP4 192.168.80.120^M
s=session^M
c=IN IP4 192.168.80.120^M
t=0 0^M
m=audio 12592 RTP/AVP 0 8 18 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:18 G729/8000^M
a=fmtp:18 annexb=no^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=silenceSupp:off - - - -^M
Incoming Egress Response
SIP/2.0 180 Ringing^M
Via: SIP/2.0/UDP 192.168.80.146:5080;branch=z9hG4bK16409b6ecd0e22515d2d247448c33bf7;rport^M
From: <sip:2408150@system;transport=udp>;tag=TiBriA^M
To: <sip:2258100@mitel1.tango-networks.com;transport=udp>;tag=0_590921104-91056377^M
Call-ID: cf2412c3-f200-4379-a38a-7947b2fc44f6^M
CSeq: 100 INVITE^M
Contact: <sip:2258100@192.168.80.143:5060;transport=udp>^M
Server: Mitel-3300-ICP 10.2.0.26_2^M
Content-Length: 0^M
^M
Outgoing Ingress Response
SIP/2.0 180 Ringing^M
Via: SIP/2.0/UDP 192.168.80.120:5060;branch=z9hG4bK53c653ea;rport=5060^M
From: "3CX admin" <sip:2408150@192.168.80.120>;tag=as2d832122^M
To: <sip:2258100@192.168.80.146:5080>;tag=0_590921104-91056377^M
Call-ID: 499274757b8ca33243049f5379eb9d70@192.168.80.120^M
CSeq: 102 INVITE^M
Contact: <sip:192.168.80.146:5080;abrazohostnode=192.168.80.146;transport=udp;lr>^M
Content-Length: 0^M
^M
Incoming Egress Response
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 192.168.80.146:5080;branch=z9hG4bK16409b6ecd0e22515d2d247448c33bf7;rport^M
From: <sip:2408150@system;transport=udp>;tag=TiBriA^M
To: <sip:2258100@mitel1.tango-networks.com;transport=udp>;tag=0_590921104-91056377^M
Call-ID: cf2412c3-f200-4379-a38a-7947b2fc44f6^M
CSeq: 100 INVITE^M
Contact: <sip:2258100@192.168.80.143:5060;transport=udp>^M
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE^M
Content-Type: application/sdp^M
Server: Mitel-3300-ICP 10.2.0.26_2^M
Content-Length: 170^M
^M
v=0^M
o=- 546 546 IN IP4 192.168.80.235^M
s=-^M
c=IN IP4 192.168.80.235^M
t=0 0^M
m=audio 50082 RTP/AVP 0 101^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=ptime:80^M
Outgoing Ingress Response
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 192.168.80.120:5060;branch=z9hG4bK53c653ea;rport=5060^M
From: "3CX admin" <sip:2408150@192.168.80.120>;tag=as2d832122^M
To: <sip:2258100@192.168.80.146:5080>;tag=0_590921104-91056377^M
Call-ID: 499274757b8ca33243049f5379eb9d70@192.168.80.120^M
CSeq: 102 INVITE^M
Contact: <sip:192.168.80.146:5080;abrazohostnode=192.168.80.146;transport=udp;lr>^M
Allow: REGISTER,REFER,NOTIFY,SUBSCRIBE,OPTIONS,PRACK,INFO,ACK,CANCEL,BYE,INVITE^M
Content-Type: application/SDP^M
Content-Length: 170^M
^M
v=0^M
o=- 546 546 IN IP4 192.168.80.235^M
s=-^M
c=IN IP4 192.168.80.235^M
t=0 0^M
m=audio 50082 RTP/AVP 0 101^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=ptime:80^M
-
I can send you a How To.
Don't want to post it since it isn't mine.
PM me with your email
Ralph
-
I would have to do some digging but looking at your captures it is listing all options that it can talk for rtp the first one will take precedence ie U Law. I believe 200s don't show as they are events not audio traffic or an sdp. If you have RTP traffic it should show the codec being used. Is the problem no audio, call fails, or just curious?
-
Do you have compression licenses and available DSP ports?
-
The 2 way audio is there.
Sometimes I hear noise on Mitel phones (sometimes OK).
I have wireshark set up on the box running softphones.
Any idea on how to run wireshark from Mitel 3300?
Checked my license file:
In Trunking/Networking section
Compression License: 0
Does that mean G.711 is disabled?
If I change from zone 1 to zone 2 (which has compression set to YES), I can see G.729 in SDP of 200 OK. But, Asterisk ends the call immediately because it refuses to do transcoding with G.729.
-
Compression licenses are only needed if you are doing T.38 and a few other things. It will use G711 by default. There is no way to wireshark on the Mitel but you could mirror the controller port in the switch to another port and capture that way.
-
I also looked at a regular sip call since I do not have an asterisk box anymore and wireshark does show G.711 ULaw Media Type with 101 Media Attribute telephone event on 200 OK.
-
You mean from 3300?
-
Was a normal call to an itsp.
-
Call from Mitel to itsp? So 200 OK is from itsp to Mitel, right?
I guess in my case the codec used between Mitel and Asterisk is Ulaw, since I recorded my MOH and AA in ULaw and I can hear them OK from my softphone.
-
Yes that is correct the point being Media Attribute is as you are seeing of Telephone-Event but does show type in wireshark as G711 ULaw.
-
Actually you do need compression licenses if you are doing G.729 out SIP trunks.
If you want to wireshark the Mitel audio you can turn the encryption off in systems options.
-
From my reading he was speaking of running it from the Mitel platform as he can run from the softphone platform. And yes I meant to say G729/T.38 but the OP was looking for G711 so the licenses are unneeded a dsp typically would not be needed either unless more resources are needed on the system.
-
Before you spend too much time on this you may want to check that Mitel actually support this config...
-
Your right it was the first line in the post
From the PBX in the middle can you do a wireshark and see if you are getting audio from both sides?
-
Yeah that is really the only way to look at sip signalling as you can see the whole packet at lot of PBX try building this in but they only show certain items and in a summary way.