Troubleshooting VoIP's Most Common SIP Problems
Failover routing for business continuity, reduced telephony costs, boosted agent productivity, and smoother communications with international customers… There are a multitude of reasons you switched over to a cloud-based system.
But like any new technology endeavor – there is a learning curve with plenty of hiccups along the way. From implementing SIP security protocols to fixing downtime and unreliable connections, you might be feeling some growing pains.
Below, we’re covering the top SIP trunking issues and how to troubleshoot them.
Troubleshooting VoIP’s Most Common SIP Problems
While setting up your SIP trunk will likely be free of charge, internal hours and resources will likely go into proper configuration. From bandwidth issues and codec mismatches, there are a variety of mishaps that can occur while switching your providers. Here are some of the most common SIP problems and tips for troubleshooting them.
1. Pairing phone systems
Not every SIP is compatible with every PBX system and even if they are, there may be licensing requirements causing issues for you.
To troubleshoot configuration issues:
→ SIP handshaking: The first thing to do when trying to identify a compatibility or SIP handshaking issue is to set up a packet capture between the two session border controllers (SBCs). From there, analyze the communications and monitor for SIP response codes. This code will help you determine the issue and proper resolution.
→ End-point testing: VoIP SIP trunks can be used across a variety of endpoints from IP phones, mobile phones and softphones. Test each endpoint to ensure your firewalls and servers aren’t blocking any traffic.
2. Choppy, lag and jittery calls
If you’re experiencing inadequate audio call quality with your SIP trunking service, it could be the result of insufficient bandwidth, network issues or limitations to SBCs. There are several things that may be causing these issues:
- Jitter: Sometimes, packets take different paths than intended which can result in scrambled audio. Sound familiar? You may be experiencing jitter, which is one of the most common SIP trunking quality issues.
- Latency: Experiencing lags in audio? How about an echo effect? You likely have a latency issue.
To troubleshoot call quality issues:
→ Quality of service: Quality of service (QoS) is a router setting that favors voice traffic over data. If it isn’t properly configured, activities that require high-bandwidth, such as downloading large files, can cause poor call quality. If your router doesn’t have the capability for QoS, it’s recommended you consider an upgrade in order to get the most out of your SIP trunking service.
→ Codecs: Next, check your PBX codec options. A codec is an algorithm used to compress audio or video which transmits high-quality media content over the internet without using large amounts of bandwidth.
To determine how much bandwidth your call center requires to maintain high quality calls, use this simple formula: Number of Concurrent Calls x 85 kbps = Required Bandwidth
Some codecs require more than 85 kbps per call, but most SIP providers use the G.711 and G.729 codecs, which usually require anywhere from 24-80 kbps to handle a single call.
3. Maintaining concurrent calls
Another SIP issue you may be experiencing is a deficient number of SIP channels. Not having enough SIP channels will lead to call issues because it limits the number of employees to make or answer calls.
So, how many channels do you need? Each channel can host one phone call at a time. Call centers or other organizations that typically experience high call volumes will need more trunks, more channels and greater internet bandwidth to support each one.
To troubleshoot SIP channel issues:
→ Track your peak call volume: Fixing this issue is actually quite simple. Simply document the number of simultaneous calls (inbound and outbound) your organization takes during peak hours. If you get 50 simultaneous calls, you will require 50 channels. Keep in mind that if you have a higher call volume, more bandwidth and channels are required.
4. Other common SIP issues
If none of these larger issues seem to hit the nail on the head, it may be a much smaller and easier issue to fix! Troubleshoot the following:
To troubleshoot other common SIP issues:
→ Do you have enough bandwidth?
This is a super common issue in regard to SIP trunking issues. Without enough bandwidth, you could experience slowed down applications and internet connection as well as degraded call quality.
As your organization continues to expand its unified communications, it will inevitably need more bandwidth to support your applications. If you haven’t upgraded in a while, simply call your internet provider to discuss upgrade options. This is a simple fix to your issue.
→ Do you have a codec mismatch?
If you can’t hear callers on the other end, you may have a codec mismatch. This is because in order to achieve two-way audio, each side of the VoIP call must exchange real-time transport protocol (RTP) within the same codec.
If you have a codec mismatch, call your SIP provider. They will be be able to correct the issue and match up your codecs accordingly.
→ Are you waiting for your numbers to port?
SIP trunking makes it possible for you to keep your existing communications infrastructure and phone numbers. And while this is a huge benefit, it can typically take a couple of weeks for all your phone numbers to transfer over.
If it’s taking longer than expected, work with your carrier and SIP provider to speed up the process or find a temporary solution while you wait.
The Benefits of SIP Trunking Outweigh The Problems
Implementing new telecommunications software comes with learning curves and like most new tools, you have to work out a few kinks in the beginning. Luckily, most of these issues are quickly and easily resolved by following the SIP troubleshooting steps listed above.
What’s more, SIP trunking service providers like AVOXI have your back each step of the way. And with our unlimited channels per SIP trunk, you can do more when you migrate your PBX to the cloud. It’s easy to see why hundreds of companies use us to help secure SIP connections and global connectivity.
|Codec Name||Bitrate per Second||Avg. Bandwidth Required||Description|
|G 729||8 Kbps||24 Kbps||Uses an algorithm for extreme compression, works well with low bandwidth|
|G 711||64 Kbps||80 Kbps||Offers lossless compression to reduce bandwidth needs|
|G 722||64 Kbps||80 Kbps||High quality but requires more bandwidth|
|G 726||24-32 Kbps||56 Kbps||Used in international SIP trunking|
|G 728||16 Kbps||32 Kbps||Offers toll voice quality for lower bandwidths|
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