Messages

1

MiVoice Office 250/Mitel 5000 / Mitel CS5000 - Latency/Jitter table

« on: October 09, 2014, 04:34:25 AM »

Hi Guys,

I'm trying to find an "official" guide from Mitel regarding latency/jitter requirements. and even better if there was some table that would identify what call quality we would expect on certain RTT ranges?

Thanks
Skip

2

MiVoice Office 250/Mitel 5000 / Re: SDP Options - CS5000

« on: April 08, 2014, 07:40:01 AM »

Hi Teach, I only have read access to the Mediant as its managed by the SIP provider. Are you able to tell me what exactly your looking for and I'll try and get the config.

Thanks

3

MiVoice Office 250/Mitel 5000 / Re: SDP Options - CS5000

« on: April 07, 2014, 09:16:50 PM »

Here is the Processor and Expansion Module config

4

MiVoice Office 250/Mitel 5000 / Re: SDP Options - CS5000

« on: April 07, 2014, 09:12:52 PM »

Hi Tech,

Your correct about the AudioCodes Mediant 1000 is not support by Mitel. The Mediant is a SIP gateway. The reason for the Mediant though is because we will be delivering SIP to our Mitel CS5000 AND also a Lync enterprise voice environment. The Mediant allows the flexibility for us to reroute the SIP calls as we see fit.

If you have a look at the attached packet capture (This was taken today)

The INVITE packet is send from 10.31.1.3:5060 (Mitel Processor Card) to 10.31.1.4:5060 (Mediant 1000). In this packet in the L7 headers its saying to use 10.31.1.2:6846 (Mitel Processor Expansion Card) for RTP traffic.

Then when the first RTP packet is sent from 10.31.1.4:6340 (Mediant 1000) to the 10.31.1.2:6846 (Mitel Processor Expansion Card) a ICMP packet is send back saying Port Unreachable... So its like the PEC hasnt opened up port 6846, which was sent in the original INVITE packet.

Thanks

5

MiVoice Office 250/Mitel 5000 / Re: SDP Options - CS5000

« on: April 04, 2014, 07:58:24 AM »

Hi Tech,

We are having problem getting SIP to work from an AudioCodes 1000 to our CS5000. (See my other post. http://mitelforums.com/forum/index.php/topic,5037.0.html)

When making a SIP call...

- The CS5000 sends an Invite packet to the AC1000.
- The AC1000 sends 'Trying' packet , then 'Ringing'
- RTP is flowing from AC1000 to the CS5000, but to UDP ports that are not ready, as the 5000 is not happy that a call has started.
- The AC1000 processes the call, then sends back a '200 OK' packet when the call is answered at the remote end.
- The 5000 does not 'ACK' the '200 OK'.
- The AC1000 ends the call, due to no response from the CS5000.
- The 5000 responds with a '400 Not found' packet , indicating that there is no call to end anyway.

There seems to be a mismatch with the SDP between the Audio Codes Gateway and the CS5000. In doing research one of the fixes for this on the Mitel 3300 is to disable the Mitel proprietary SDP. However looking at the CS5000 I'm unable to find any SDP settings, hence my reason for asking.

Thanks

6

MiVoice Office 250/Mitel 5000 / SDP Options - CS5000

« on: April 02, 2014, 11:08:05 PM »

Hi Guys,

Is there a way to change the SPD options on a CS5000(V5.1.0.46), like on the Mitel 3300?


We're trying to troubleshoot a SIP Trunk problem where the SDP/SIP Headers are acting up.

http://britishschoolsofamerica.org/uwi/help/En/sysadmin/forms/sip_peers.html

Thanks

7

MiVoice Office 250/Mitel 5000 / Re: Invite Packet Port Closed - SIP Call

« on: March 29, 2014, 08:35:06 PM »

Hi Marcolive - See attachted.

8

MiVoice Office 250/Mitel 5000 / Invite Packet Port Closed - SIP Call

« on: March 28, 2014, 03:11:15 AM »

Hi All,

We recently got SIP trunks installed and in the process of setting these up we've come across a problem when making a call through the SIP trunks. An outgoing call through the SIP trunk is able to be established, but once established there is no speech and then after 5-10 seconds the call is then dropped.

Intertel CS5000 5.1.0.36
Processor Card 10.31.1.3
Processor Expansion Card 10.31.1.2
Audiocodes Mediant Gateway 10.31.1.4

Having a look at the packet capture the SIP invite packet from the Processor Card (10.31.1.3) sends the connection information for RTP packets as 10.31.1.2, port 7218. Then when the Mediant Gateway (10.31.1.4) starts sending RTP packets to 10.31.2, port 7218 as stated from the SIP Invite packet an ICMP reply is send from the Process Expansion Card (10.31.1.2) back to the Mediant Gateway (10.31.1.4) saying that the port is closed... See attached for the packet captures.

So what I can tell is that the PC is either sending out the wrong port number or the port isnt opening on the PEC. Each time a new call is made the RTP port changes, but its the same outcome.


Any help would be greatly appreciated.

Cheers
Andrew