VCM issues - SIP to 96xx handsets.

[shardian]

Hi - hope someone can help me out here.
I have an IP500 running 8.0(43). System has a phone8 and a VCM32 in system and uses both lan ports. LAN1 is used for internal IP extensions. Address is 192.168.42.1. LAN2 is used for the customer's SIP trunks. Address is 192.168.1.251.

There are a mix of 9620's and 9608's as well as some R4 DECT handsets (and bases). All handsets are running the latest firmware.

The issue I'm having is that when a call is put on hold by any of the IP handsets, the external party doesn't get any audio - just silence. If the DECT handsets or any of the analogue extension put anyone on hold, the music plays. I should also mention the hold music is a file on the SD card.

After looking into it, it appeared that regardless of Direct Media Path being on or off for the IP phones, if the SIP trunk and the handset were able to use the same codec, RTP relay was used for the connection type.

My workaround was to make all the trunk use one codec and all the IP phones to use another codec. Testing this worked fine and the hold music played. Not believing that it could be that easy, I tested a bit more and found that not every inbound call would work as expected. Every call rings as its meant to, but sometimes when the call is answered, the caller continues to get ringback tone but the phone that has answered the call gets silence. Status shows the call as answered, but its all a bit odd.

Has anyone seen anything similar and found a fix?
Also - I've swapped the VCM 3 times to ensure it's not a faulty board and tested this on another chassis in the office to prove ensure its not faulty equipment on site.

Thanks for your help.

-Matt

 

[tlpeter]

What is your IProute in the IPO?
It sounds like RTP is not always reaching the SIP trunk on the other end.


BAZINGA!

I'm not insane, my mother had me tested!

 

[hairlessupportmonkey]

Or perhaps a flakey STUN server if you are using one?

ACSS - SME
General Geek

 

[shardian]

Hey guys.

Thanks for the replies. Not using STUN and I've tried creating routes for every IP address we'd be getting traffic from/sending to. I've also created a default route for any traffic. Same thing happens both times.

Handset to handset obviously uses RTP relay and the hold music plays fine. Just not when talking to an external party.

Any other ideas?

 

[tlpeter]

You use sip you say.
Can you try something? go to the sip trunk and see if the re-invite is turned on or off.
Turn it on or off (what ever it is now, try the opposite)

We have had issues with complete silence after the first announcement and there was also not tone or MOH anymore.


BAZINGA!

I'm not insane, my mother had me tested!

 

[shardian]

Tried that - same thing happens.

When I have different codecs for the trunks vs the extensions, so the VCM card 'controls' the call - I still have weird issues with calls appearing answered, but still getting ring back tone at the caller's end.

When the same codec is in use for both trunks and extensions and the connection is RTP relay, the call is able to be answered and 2 way audio works, however the external party doesn't hear MOH.

 

[tlpeter]

Does your IPDECT has a different default gateway then your phones?
I have seen issues with no speech when the default gateway is not correct.


BAZINGA!

I'm not insane, my mother had me tested!

 

[shardian]

Just an update with this one.

I swapped the router we were using out and replaced it with a Cisco SRP model and all is fine.

From what I can tell with my limited Wireshark knowledge, when an analogue device or one of the IP DECT phones put someone on hold, the RTP stream back to the ITSP is sent through the router without any problems. However when an IP set places a call on hold, its RTP stream to the system is suspended and it seems that the router won't continue to allow the RTP stream from the system to ITSP.

I've tried a few different model routers since (mostly just small business models) but the Cisco is the only one that seems to work with this setup.

Thanks for the all the suggestions.