You have deployed a VoIP solution and you are having quality issues. So, whether the communications must be of a quality equivalent to RNIS or close (in the case of compression), you experience blanks, echoes, a metallic voice or even communication cuts.
In the article, I will consider that the site has sufficient quality computer cabling and that the deployed equipment is of good quality (switch, router, IPBX, telephone sets, gateways, etc.).
Determine the origin of the problems
It will be important first to determine if the problem is with external communications or if the problem also occurs with communications on the same site. If the calls between 2 IP stations (important) on the same site (with the local IPBX if we are not in centrex) are degraded, this problem will have to be resolved first. For this, it is necessary to check some crucial parameters:
- which codec is used: the codec has an influence on the perceived quality. if you have a PABX, use the G711 or even G722 (wideband codec for better voice reproduction). In centrex mode it may be desirable to save bandwidth, so the G729 will be the codec of choice.
- peer to peer mode activated : if so, the RTP stream remains on the local network in centrex mode, the connection being made from peer to peer. The WAN therefore does not intervene in the voice flow apart from the signaling (which can be considered at first as negligible from a bandwidth point of view when compared to RTP flows). Otherwise, for a call between 2 stations on the same site, we have 1 outgoing call and 1 incoming call on the site, so there is an impact on the WAN.
- is VAD (Voice Activity Detection) activated ? this interesting feature allows you to limit the bandwidth of a communication by removing packets including silences. Depending on the settings, the savings may be significant. This function is useless locally, and must be deactivated. It is difficult to set up properly, and in the event of an error the words will be cut, communications may become inaudible.
- has a voice VLAN been set up applying voice CoS to the concerned flows?
- check that the equipment is up to date, because a firmware version may have a defect explaining the problem.
In general, the problems are mostly encountered during external calls. Indeed, the bandwidth is then more limited, communications sometimes pass over the Internet and the link is also sometimes shared with other flows.
VoIP via internet
I cannot recommend enough not to carry voice over the internet. The Internet was not designed to carry real-time streams and cannot guarantee quality (that doesn’t mean it doesn’t work). In addition, security will also be difficult to ensure. I let you imagine what I think of the SIP trunk offers via the internet. One need only take a look at the list of security vulnerabilities to be afraid, but that’s not the topic of the day.
Other essential information
In addition to the questions asked above, you must also obtain other information:
- Number of simultaneous external calls
- Broadband link bandwidth (in IP)
- Is the link dedicated to voice? if not, what are the mechanisms that prioritize voice over other streams?
- Do voice streams use the internet or stay on the operator’s network?
- By measuring the flows on the link, you will be able to check if the link is saturated, if the rules are applied correctly, or if the link is suffering from packet loss or jitter.
Not knowing your particular cases, I can’t go much further, but if you have sufficient networking knowledge and answering the questions below, you have all the balls to find the source of the problem. To help you a little more, one can get some interesting information by listening to the voice quality.
- You have blanks short of communication, you have to look at the VAD side if it is activated. Then, this effect may be caused by packet loss or excessive jitter of the WAN link. It can also be due to an ill-defined QoS rule.
- You have noises or crackles. They are often caused by LAN or WAN congestion.
- You have a robot voice, look at packet loss due to excessive jitter. (remains to find the source of this jitter).
- You echo. This is due to a problem with the echo canceller setting of the PABX or the gateway.