Does anyone know if an MTP is required for a CME to send calls to a SIP trunk provider? I have two CME's (version 4.3. and a version 7.0). They are configured identically with the exception that I registered a transcoder on the 7.0 CME. Now, my question is, is there something changed in the newer version 7.0 that requires an MTP be registered to itself? The call connects, but the distant end cannot hear me and DTMF is not working.
No. CME always sends media for external calls to the gateway.
That doesn't exclude NAT, firwall, config and other possible probelms.
Fo safe results recommend you don't use transcoder, rather G.711u everywhere.
Thats the weird part. There is no firewall involved here. This router has a public IP on the WAN interface and it goes straight out to the internet. I put the transcoder on there in case it was needed, but right now it's doing g711ulaw on the phones and trunk. Now, I am NAT'ing all the internal traffic, but it's a rather simple NAT config.
Have you configure sip bind under the external interface ?
You can check the IP addreses being used for media with show call active voice.
Note that after you will have it working, you risk almost sure toll fraud if you don't restrict SIP access to the device.
Main role of MTP is to negotiate DTMF. If you have an ability to configure DTMF as per service provider's need then usually MTP is not needed. MTP also takes care of Re-Packetization ( Codec Conversion ) of a Stream i.e. u-law to a-law conversion & vice versa.
If the trunking has above needs then yes MTP does the job & hence is needed.
Hope it helps ?
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Actually, no, I haven't enabled an MTP yet. DTMF does work now because of my provider. I was able to get through some prompts on an automated system now. However, for example, I am calling a friend that answers after I go through a few prompts, he can't hear me. I can hear him, but not the other way around.
OK, I'll give it a shot. Yeah, 100% sip. The dial-peer points to the sip-ua config where my authentication/registrar/credentials/sip-server config is.
Again, MTP is absolutely NOT required with CME, for the reasons expalined above.
Pande: can you proivide a link where Cisco says to use MTP with CME ?
OK, did a sh call active voice during the call and the only thing I can find in reference to an IP besides the ip of my sip provider is a line labeled "protocolcallid". That line has a bunch of random characters in it but after the characters is
firstname.lastname@example.org..X.X <<--- My public IP.
Apart from the internal discussions I found ( where use of MTP is confirmed but those discussions are not accessible outside Cisco ) I could find one bug which does indicate towards the MTP use.
This bug is for the SIP trunk connection between CME & CUCM but if you replace CUCM with PSTN, the same logic applies there too.
I'll continue my search for a public facing document ( sadly, CME SRND & Admin guide are not clear enough on it ) & will post it here.
Most DTMF internetworking issues will be resolved by the use of an MTP. You would need to troubleshoot how your phones is sending DTMF tones to the CME to actually tell if the MTP would be required for your environment or not. However we see a tendency to an increase on MTP usage on most SIP deployments if it's a multiprotocol setup.