Change default media port

[Akash Thimmaiah]

Hi there,

Recently have purchased a SIP Trunk and the SIP connection is established from Mitel 3300 (MiVoice Business) to SIP Server. Here the issue is unable to hear audio when inbound and outbound call is placed. According to our findings, the packets flowing through rtp port for media communication of SIP provider uses port range between 20000-24000 but our Mitel 3300 PBX use the port range between 9000-9001.

So can we modify the Mitel 3300 pbx RTP port?


Regards,
Akash

 

[techymitel]

UDP port 9000 was used for 5550 IP console voice. If you are using 5550 IP consoles then RTP will be 9000 (channel 1) and 9002 (channel 2).

What software level in the MiVoice Business?

What kind of handsets are you using?

Do you have MBG for SIP proxy?

If there is no MBG and you are using MiNet handsets, then it's likely the handsets will expect RTP to be sent to them on ports 50000-505111

If there is no MBG, the SIP media server will need to be able to stream voice directly to the handsets.

You could enable and check the voice quality stats to see if you are experiencing packet loss.

 

[Akash Thimmaiah]

Hi techymitel

Here is the answer for your question

What software level in the MiVoice Business?
Release level: 8.0 SP2

What kind of handsets are you using?
I am using 6920 and Micollab softphone (Micollab app)

Do you have MBG for SIP proxy?
No, I dont have a standalone MBG.

Currently I see the the packets as follows
]INVITE sip:9999xxxxxx@siptrunk.net2phone.com:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1573778720-106010907
Route: <sip:siptrunk.net2phone.com:5060;transport=udp;lr>
Max-Forwards: 26
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: "Access Online" <sip:121111xxxx@siptrunk.net2phone.com>;tag=0_1573778720-106010908
To: <sip:9999xxxxxx@siptrunk.net2phone.com>
Call-ID: 1573778720-106010906
CSeq: 1 INVITE
Contact: "Access Online" <sip:121111xxxx;tgrp=SIP;trunk-context=siptrunk.net2phone.com@192.168.1.2:5060;transport=udp>
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 14.0.2.26
P-Asserted-Identity: "Access Online" <sip:121111xxxx@siptrunk.net2phone.com:5060;transport=udp>
Supported: replaces
Content-Length: 167

v=0
o=- 7494 7494 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 9000 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive

 

[Akash Thimmaiah]

Thanks techymitel, we were able to achieve it using MBG