A few weeks ago we stablished a 4MB MPLS with our VOIP provider. 50% bandwith was assigned for voice in the QoS profile. We only use this link for voice, no data at all.
When calls are routed throught internet every goes fine, quality is aceptable but when we switch to the MPLS connection we experience choppy audio for caller and callee. When MPLS is plugged, what is weird is that when we use silent monitor for listenig into calls, we hear that local calls are choppy eventhought this is supposed to be inside the LAN.
Our MPLS provider has done all necesary tests to determine that there are no lost packets between the edge routers. We did the same for our LAN. We use the same LAN for both links. this is not a bandwith issues because we are running under 40% of the bandwith capacity.
We did a test with an Audio Codes SIP Media Gateway (we used g711) connected directly to the MPLS Router and sound quality is fine, no choppiness. To discard that our LAN was causing the problem, we did the same test connecting the Asterisk straight to the MPLS router and a Snom IP Phone directly to the Asterisk box and audio was choppy (using both g711 and g729).
So my gues is that the problem is in the box.
Freepbx 2.5 - Asterisk 1.4
Xeon Dual Core - 4 GB RAM 120 GB HD
I appreciate any help.
This is truly an oddball condition. My money’s on a flaky network interface in the asterisk box. From the perspective of devices on your network, they have no idea their traffic crosses an MPLS backbone - that’s the whole point of MPLS - the customer side, from the CE router on in have no idea what’s up in the network.
Are you sure the provider checked the ingress and egress PEs to make sure the CoS parameters are being applied, traffic gets queued and forwarded correctly, etc.? RTP traffic should already be marked for Expedited Forwarding.
Yes, I checked with the provider and CoS paramenters are being applied, I got the conclusion that the problem is residing in the asterisk box because we replaced the Asterisk box for an Audio Codes SIP media gateway connected straight to the MPLS router, did same tests and voice quality was good, no choppines at all.
I too have seen a full- and half-duplex mismatch between the asterisk NIC and the switch, or any other two adjacent network device reek havoc on audio quality. This duplex mismatching can be especially true if 10base-T port(s) are involved.
Keeping in mind that the asterisk box is not the only place this could be askew; therefore, verify your actual vs. expected along the entire ethernet path, especially at the adjacencies that changed as a result of the MPLS CE router insertion.
# ethtool eth0
Settings for eth0:
Supported ports: [ TP MII ]
Supported link modes: 10baseT/Half 10baseT/Full
Supports auto-negotiation: Yes
Advertised link modes: 10baseT/Half 10baseT/Full
Advertised auto-negotiation: No
Supports Wake-on: g
Current message level: 0x00000007 (7)
Link detected: yes
I always turn auto-negotiation off, hard-set the speed and duplex.
# cat /etc/sysconfig/network-scripts/ifcfg-eth0
# Intel Corporation 82801BA/BAM/CA/CAM Ethernet Controller
ETHTOOL_OPTS="speed 100 autoneg off duplex full"
I suggested checking your port negotiation in response to your similar query on the trixbox forum.
I also suggested that you check for fragmentation as MPLS wrapper can cause fragmentation on large packets.
For fragementation checking you will have to run wireshark. ethtool and mii-tool will both help you with link layer issues.
Hi! Since I am interested on doing a project using Asterisk and MPLS for my MSc course, I would like to ask: In the above scenario, is it VoMPLS or VoIPoMPLS? Could you please clarify. Sorry for hijacking the thread and for being off-topic…