SIP Trunk Misconfigurations That Kill Call Quality (and How to Fix Them)

SIP trunks are the ruling choice for enterprise voice in Australia, which helps to support not just cloud UC platforms but also on-premise PBXs. They deliver flexibility, scalability, and cost efficiency compared to legacy telephony, but they are far less tolerant of network misconfiguration. Unlike traditional voice services, SIP relies entirely on the data network’s ability to handle real-time traffic. When users report choppy audio, one-way calls, or dropped conversations, bandwidth is rarely the issue. In almost every case, call quality problems stem from how firewalls, routers, and switches are configured.

Firewall and NAT Traversal Headaches

The Problem of SIP ALG and State-Full Firewalls

SIP ALG is one of the most damaging features that many corporate firewalls still keep enabled by default. Even though it is meant to "assist" SIP traffic through NAT, it often rewrites SIP headers and SDP bodies incorrectly, thus ruining modern SIP implementations. One-way audio, failed call transfers, disconnections after answering the call, and unstable registrations are some of the consequences. Due to the firewall's unpredictable interference, these issues appear to be random. The complete disabling of SIP ALG and replacing it with an explicit, standards-based setting is the only reliable and consistent solution. You get control, and the firewall will not interfere with SIP communication through static NAT or port forwarding for RTP media and SIP signalling.

UDP Timeouts and Dropped RTP Streams

RTP (Real-time Transport Protocol) streams utilise UDP (User Datagram Protocol) for carrying actual audio, while SIP (Session Initiation Protocol) signaling establishes a connection. A considerable number of firewalls implement very short timeout values which are totally unsuitable for voice and regard UDP traffic as wasteful. When these timers reach zero, the firewall shuts down the RTP ports during the call. The users may still perceive the call as active, but the sound suddenly disappears. The conventional "sudden silence" scenario is evaded through the practices of increasing UDP session timeouts and accurately defining the RTP port range, thus ensuring media streams are available for the whole duration of the talk.

Quality of Service (QoS) and Bandwidth Mismanagement

Incorrect DSCP Marking and Trust Boundaries

Voice traffic will only be protected by QoS if it is applied uniformly across the entire network path. It is a must to tag SIP and RTP with the DSCP values that are appropriate for them, which is mainly EF for the media, albeit tagging by itself is not enough. If access switches, edge routers, or carrier handoffs distrust the indications and therefore, do not prioritise them, voice traffic will be treated just like any other data. Call quality becomes worse during congestion. It is vital to have precisely defined trust boundaries and to carry out the verification that WAN and carrier networks do support DSCP values end to end in order to have an efficient SIP trunk QoS implementation.

Jitter Buffer Misalignment and Latency

Latency and jitter are often increased by jitter buffer misalignment instead of poor network capacity. The audio quality comes out as broken or machine-like because of the use of buffers that are too small and cannot cope with the average packet fluctuation. Buffers of excessive size lead to unnecessary delays that make communication sound unnatural. Achieving interoperability in jitter buffer behaviour across telephones, softphones, gateways, and PBXs—preferably through adaptive settings—is often the way to go when quality issues are to be resolved without increasing bandwidth.

Codec Mismatch and SIP Header Errors

Inconsistent Codec Negotiation

Codec negotiation during SIP call setup using SDP may lead to mismatches that occasionally lower the quality of the call. In the case of such mismatches, when SIP trunks and PBXs suggest using either inefficient or incompatible codecs, then the transcoding process might be necessary, or the use of low-quality compression might be the only option left. This leads to a scenario where the audio is of poor quality, the delay has increased, and the CPU usage is higher. The solution is discipline: set codec priority on the SIP trunk and the PBX, typically choosing G.711 for convenience and quality and G.729 only if there really are bandwidth issues.

Malformed or Missing SIP Registration Data

The registration phase marks the beginning of several SIP issues. Among the causes of unstable registrations are wrong SIP URI formatting, lack of authentication data, differing usernames and passwords, or unusual port usage. The effects of these problems are indicated by unreliability, calls being dropped when coming in, or being unable to make outgoing calls. Making sure that SIP registration is pure, according to standards, and fully aligned with provider specifications lays a strong ground for media and signaling.

Dealing with the quality issues of SIP trunk calls is a task that requires precision at the level of the network. Bandwidth is seldom the limiting factor. The direct or indirect impacts of SIP ALG, firewall timeouts, QoS trust boundaries, jitter buffers, codec negotiation, and registration accuracy determine whether voice is clear, noisy, or chaotic. By addressing these fundamental issues one at a time, IT departments can eliminate long-term problems and restore high-quality voice services for the firm.

In case your company is still facing the issues of SIP trunk call quality, it is highly likely that the problem lies within the configuration details. A systematic inspection of media processing, signalling pathways, and firewall behaviour often leads to the root problem. For an environment audit and the deployment of effective SIP trunk call quality solutions that have been proven successful for Australian enterprise networks, get in touch with the Anticlockwise team.