# Network readiness assessment for cloud calling

Canonical: https://warmtransfer.net/knowledge/network-readiness-assessment

Last verified: 2026-09-24

A network readiness assessment establishes whether an enterprise data path meets the latency, jitter, loss, and bandwidth thresholds required for real-time cloud calling before deployment[^57][^39]. Cisco places network readiness and requirements in the Prepare phase of a Unified CM to Webex Calling migration, which comes before the Plan phase[^52].

## Network quality targets and baseline metrics

Cisco recommends network targets for voice quality over the internet consisting of 1-way latency below 100 milliseconds, jitter below 10 milliseconds, and packet loss of 0.5 percent or less[^57]. For Webex Calling calls over the internet, Cisco's migration Prepare guidance sets a 1-way latency target of less than 100 ms, a jitter target of less than 10 ms, and a packet loss target of 0.5% or less[^47][^46][^48]. The network connection between a Webex Calling Local Gateway and Webex Calling should have a maximum 1-way latency of 100 ms, a maximum packet jitter of 10 ms, and a maximum packet loss of 0.5%[^14][^13][^15].

ITU-T G.114 'One-way transmission time' is in force as the 05/2003 edition, with Amendment 1 (09/2003) adding VoIP delay guidance and Amendment 2 (11/2009)[^12].

WarmTransfer's reading of the sources is that Microsoft's current Teams network preparation page covers requirements, optimization tasks, and bandwidth, but appears to publish no numeric latency, jitter, or packet loss targets[^29]. In retired guidance for Skype for Business Online, Microsoft listed optimal client-to-Microsoft-edge targets of RTT below 60 ms, average packet loss below 0.5%, and jitter below 3 ms[^21].

## Assessment tools and testing procedures

Cisco's Prepare guidance names CScan (cscan.webex.com) as the tool for assessing network quality, estimated call capacity and latency[^40]. The CScan basic test measures download and upload bandwidth, round-trip latency, a traceroute, a subset of TCP ports and an estimate of concurrent calls[^7]. CScan measures packet loss and jitter (upload and download) only in its Advanced Diagnostic Test, not in the basic test[^6]. Because of browser WebRTC limits, CScan validates only 2 UDP ports per test rather than the full Webex Calling UDP range[^11]. Cisco describes a CScan result as a single point-in-time estimate that does not guarantee performance at peak traffic times[^10].

The Microsoft Teams Network Assessment Tool streams packets to the nearest Teams relay edge site to measure packet loss, jitter and round-trip time, and it also checks connectivity to the required IPs and ports[^28]. In retired guidance, Microsoft stated that ping (ICMP) is not an effective way to measure network performance and recommended the Network Assessment Tool instead[^19]. Microsoft's retired guidance recommended testing for at least a week, sampling every 10 minutes and comparing the 90th percentile against the targets[^22].

The Microsoft 365 admin center network assessment scores Teams from the median UDP latency, jitter and packet loss over the previous 3 days, converted to a mean opinion score, where above 87.5 points is good and below 50 is bad[^18]. In that assessment, a score of 80 is the healthy baseline, above which regular user complaints about network performance should not be expected[^17].

## Bandwidth calculation and capacity planning

Cisco sizes Webex Calling throughput as the number of expected concurrent calls multiplied by the per-call bandwidth for the codec in use[^39].

For Microsoft Teams, 1-to-1 audio bandwidth per endpoint is listed as 10/10 kbps minimum, 58/58 kbps recommended and 76/76 kbps for best performance (up/down)[^25]. Teams Network Planner (Teams admin center > Planning > Network planner) calculates bandwidth requirements per site from the sites, network details and personas an admin enters[^30]. Using Teams Network Planner requires the Global Administrator, Teams Administrator or Teams Communications Administrator role[^32]. With Microsoft's recommended personas, Network Planner automatically splits a site's users 80% office worker and 20% remote worker, and WAN and internet capacity must be entered to get bandwidth recommendations[^31].

| Platform | Metric / Calculation | Documented specification |
| --- | --- | --- |
| Webex Calling | Bandwidth throughput formula | Expected concurrent calls multiplied by per-call bandwidth for codec in use[^39] |
| Microsoft Teams | 1-to-1 audio bandwidth (minimum) | 10/10 kbps up/down[^25] |
| Microsoft Teams | 1-to-1 audio bandwidth (recommended) | 58/58 kbps up/down[^25] |
| Microsoft Teams | 1-to-1 audio bandwidth (best performance) | 76/76 kbps up/down[^25] |

## Edge, routing, and firewall preparation

Cisco says to keep the default MTU of 1500 bytes for all IP packets carrying Webex Calling and Webex Aware services[^49]. Filtering Webex Calling traffic by IP address alone is not supported because some Webex IP address pools are dynamic and can change at any time[^45]. Cisco recommends turning off SIP ALG on routers and firewalls carrying Webex Calling traffic because some implementations break firewall traversal[^55]. When a proxy is configured, only Webex Calling HTTP/HTTPS signaling goes to the proxy; SIP signaling and media do not[^53].

Microsoft recommends a split-tunnel VPN so Teams traffic bypasses the VPN, noting that VPNs are usually not built for real-time media and some do not support UDP[^36]. For Teams, Microsoft says the firewall must not change mapped NAT addresses or ports for UDP (session persistence)[^37]. Microsoft's Teams QoS guidance recommends 3 to 5 hops from client to network edge and 3 hops from the ISP to the Microsoft network edge[^26].

When the connection to Webex breaks, Webex Calling endpoints switch to Survivability mode, register with the local Survivability Gateway and keep internal, PSTN and emergency calling[^23].

## Quality of Service settings

Cisco's Prepare guidance says to implement QoS in the on-premises environment to prioritize real-time traffic and keep it in place across switches, routers and firewalls[^54]. Webex Calling assigns the following markings:
- Webex Calling marks audio from the Webex App and from devices as DSCP 46 (Expedited Forwarding)[^41].
- Webex Calling marks video from the Webex App and from devices as DSCP 34 (AF41)[^43].
- Webex Calling marks device signaling as CS3 (DSCP 24) and Webex App signaling as CS0 (DSCP 0)[^42].

For Microsoft Teams, the recommended initial QoS configuration includes:
- Microsoft's recommended starting QoS configuration for Teams audio is client source ports 50000-50019 marked DSCP 46 (EF)[^33].
- Microsoft's recommended starting QoS configuration for Teams is video on 50020-50039 at DSCP 34 (AF41) and screen sharing on 50040-50059 at DSCP 18 (AF21)[^35].

Microsoft says Teams QoS works as expected only when it is applied on every link between callers, so for remote users traffic can be prioritized only within the internal managed network[^34]. Microsoft's retired guidance states that traffic crossing the public internet to the Microsoft network does not keep QoS markings or prioritization, while ExpressRoute does[^20].

## Monitoring operational call quality

Control Hub grades a Webex Calling endpoint's experience as good only when packet loss is below 5%, latency or RTT is below 400 ms and jitter is below 150 ms[^3]. Control Hub grades a Webex Calling cloud media relay hop as good only when packet loss is below 2.5%, latency or RTT is below 200 ms and jitter is below 75 ms[^4]. In Control Hub call troubleshooting, exceeding any single threshold (loss, latency or jitter) marks the endpoint or hop as poor[^2].

WarmTransfer's reading of the sources is that Control Hub's 'good' grading limits (400 ms RTT, 150 ms jitter, 5% loss) are far looser than Cisco's readiness targets (100 ms one-way, 10 ms jitter, 0.5% loss), so a network whose calls grade good can still fail readiness[^5].

## See also

- See also [Webex Calling Site Survivability Gateway](https://warmtransfer.net/knowledge/webex-calling-survivability-gateway).
- See also [CUCM to Webex Calling migration](https://warmtransfer.net/knowledge/cucm-to-webex-calling).
- See also [QoS marking and queuing for voice](https://warmtransfer.net/knowledge/qos-marking).
- See also [SD-WAN considerations for voice](https://warmtransfer.net/knowledge/sdwan-voice).
- See also [Firewall ports and IP ranges for cloud calling](https://warmtransfer.net/knowledge/firewall-ports-uc).
- See also [Codec selection and bandwidth planning](https://warmtransfer.net/knowledge/bandwidth-codec-planning).
- See also [DNS SRV and service discovery for SIP](https://warmtransfer.net/knowledge/dns-srv-voice).
- See also [Webex Edge Connect private cloud connectivity](https://warmtransfer.net/knowledge/webex-edge-connect).

## Applicability

Applies to: Cisco Webex Calling, Cisco Webex Calling Local Gateway, ITU-T G.114, Microsoft Teams, Microsoft Skype for Business Online (retired), Cisco CScan, Microsoft Teams Network Assessment Tool, Microsoft 365 network connectivity, Cisco Webex Calling Survivability Gateway, and Cisco Webex Calling (Control Hub troubleshooting). Deployments: multi-tenant and standard. Sources checked 2026-09-24.

## What remains uncertain

A current Microsoft source for Teams latency jitter loss targets is not covered by the sources below. Published readiness targets for other UCaaS platforms are not covered by the sources below. Webex Calling specific content in Network Requirements for Webex Services is not covered by the sources below. The network chapter of the Preferred Architecture for Webex Calling PDF is not covered by the sources below. Direct Routing SBC network and media requirements for Teams Phone readiness are not covered by the sources below. A raw-HTML check of the order and wording of the Cisco Prepare network readiness checklist and bandwidth table rows is not covered by the sources below. The Teams Network Assessment Tool usage guide and its built-in media quality targets are not covered by the sources below. Whether the live CScan UI applies numeric pass or warn thresholds is not covered by the sources below.

## Sources

[^1]: Control Hub call troubleshooting shows Webex Calling calls from the past 21 days. Source: [Troubleshoot Webex Calling calls in Control Hub](https://help.webex.com/en-us/article/frj1efb/Troubleshoot-Webex-Calling-Media-Quality-in-Control-Hub), search / call list instructions. Checked 2026-09-24.
[^2]: In Control Hub call troubleshooting, exceeding any single threshold (loss, latency or jitter) marks the endpoint or hop as poor. Source: [Troubleshoot Webex Calling calls in Control Hub](https://help.webex.com/en-us/article/frj1efb/Troubleshoot-Webex-Calling-Media-Quality-in-Control-Hub), section 'Hop details view'. Checked 2026-09-24.
[^3]: Control Hub grades a Webex Calling endpoint's experience as good only when packet loss is below 5%, latency or RTT is below 400 ms and jitter is below 150 ms. Source: [Troubleshoot Webex Calling calls in Control Hub](https://help.webex.com/en-us/article/frj1efb/Troubleshoot-Webex-Calling-Media-Quality-in-Control-Hub), section 'Hop details view', end-to-end experience thresholds. Checked 2026-09-24.
[^4]: Control Hub grades a Webex Calling cloud media relay hop as good only when packet loss is below 2.5%, latency or RTT is below 200 ms and jitter is below 75 ms. Source: [Troubleshoot Webex Calling calls in Control Hub](https://help.webex.com/en-us/article/frj1efb/Troubleshoot-Webex-Calling-Media-Quality-in-Control-Hub), section 'Hop details view', media relay point thresholds. Checked 2026-09-24.
[^5]: Control Hub's 'good' grading limits (400 ms RTT, 150 ms jitter, 5% loss) are far looser than Cisco's readiness targets (100 ms one-way, 10 ms jitter, 0.5% loss), so a network whose calls grade good can still fail readiness (inferred). Source: [Troubleshoot Webex Calling calls in Control Hub](https://help.webex.com/en-us/article/frj1efb/Troubleshoot-Webex-Calling-Media-Quality-in-Control-Hub), section 'Hop details view', compared with cisco-help-4yz2zd-prepare-migrate-ucm-wxc 'Network readiness and requirements'. Checked 2026-09-24.
[^6]: CScan measures packet loss and jitter (upload and download) only in its Advanced Diagnostic Test, not in the basic test. Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), section 'Advanced diagnostic test'. Checked 2026-09-24.
[^7]: The CScan basic test measures download and upload bandwidth, round-trip latency, a traceroute, a subset of TCP ports and an estimate of concurrent calls. Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), section 'Basic test'. Checked 2026-09-24.
[^8]: CScan cannot detect whether the network is wired or wireless, DHCP or DNS availability, SRV support, NTP, proxies, QoS or VPN. Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), section 'Before you begin' (items CScan does not test). Checked 2026-09-24.
[^9]: The CScan help article does not appear to publish numeric pass/fail thresholds for latency, jitter or loss, so a CScan result has to be compared against Cisco's separately published targets (inferred). Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), whole article; sections 'CScan report information' and 'Interpret test results'. Checked 2026-09-24.
[^10]: Cisco describes a CScan result as a single point-in-time estimate that does not guarantee performance at peak traffic times. Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), section 'Interpret test results'. Checked 2026-09-24.
[^11]: Because of browser WebRTC limits, CScan validates only two UDP ports per test rather than the full Webex Calling UDP range. Source: [Use CScan to test Webex Calling network quality](https://help.webex.com/en-us/article/y27bej/Use-CScan-to-Test-Webex-Calling-Network-Quality), section 'Basic test' (TCP/UDP ports note). Checked 2026-09-24.
[^12]: ITU-T G.114 'One-way transmission time' is in force as the 05/2003 edition, with Amendment 1 (09/2003) adding VoIP delay guidance and Amendment 2 (11/2009). Source: [G.114: One-way transmission time (recommendation page)](https://www.itu.int/rec/T-REC-G.114), recommendation listing, in-force entries. Checked 2026-09-24.
[^13]: The network connection between a Webex Calling Local Gateway and Webex Calling should have a maximum packet jitter of 10 ms. Source: [Get started with Local Gateway](https://help.webex.com/en-us/article/t9xctu/Get-started-with-Local-Gateway), section 'Connection qualifications'. Checked 2026-09-24.
[^14]: The network connection between a Webex Calling Local Gateway and Webex Calling should have a maximum one-way latency of 100 ms. Source: [Get started with Local Gateway](https://help.webex.com/en-us/article/t9xctu/Get-started-with-Local-Gateway), section 'Connection qualifications'. Checked 2026-09-24.
[^15]: The network connection between a Webex Calling Local Gateway and Webex Calling should have a maximum packet loss of 0.5%. Source: [Get started with Local Gateway](https://help.webex.com/en-us/article/t9xctu/Get-started-with-Local-Gateway), section 'Connection qualifications'. Checked 2026-09-24.
[^16]: Certificate-based Local Gateway trunking needs a public IP or static NAT, while dynamic NAT is preferred for registration-based trunking because it needs fewer firewall changes. Source: [Get started with Local Gateway](https://help.webex.com/en-us/article/t9xctu/Get-started-with-Local-Gateway), prerequisites for certificate-based and registration-based trunks (firewall and NAT requirements). Checked 2026-09-24.
[^17]: In the Microsoft 365 network assessment, a score of 80 is the healthy baseline, above which regular user complaints about network performance should not be expected. Source: [Microsoft 365 network assessment](https://learn.microsoft.com/en-us/microsoft-365/enterprise/office-365-network-mac-perf-score), introduction and expected user experience table. Checked 2026-09-24.
[^18]: The Microsoft 365 admin center network assessment scores Teams from the median UDP latency, jitter and packet loss over the previous three days, converted to a mean opinion score; above 87.5 points is good and below 50 is bad. Source: [Microsoft 365 network assessment](https://learn.microsoft.com/en-us/microsoft-365/enterprise/office-365-network-mac-perf-score), section 'Microsoft Teams'. Checked 2026-09-24.
[^19]: Microsoft's retired guidance says ping (ICMP) is not an effective way to measure network performance and recommends the Network Assessment Tool instead. Source: [Media Quality and Network Connectivity Performance - Skype for Business Online (retired)](https://learn.microsoft.com/en-us/previous-versions/skypeforbusiness/optimizing-your-network/media-quality-and-network-connectivity-performance), section 'Measuring network performance', Note. Checked 2026-09-24.
[^20]: Microsoft's retired guidance states that traffic crossing the public internet to the Microsoft network does not keep QoS markings or prioritization, while ExpressRoute does. Source: [Media Quality and Network Connectivity Performance - Skype for Business Online (retired)](https://learn.microsoft.com/en-us/previous-versions/skypeforbusiness/optimizing-your-network/media-quality-and-network-connectivity-performance), section 'Implementing Quality of Service (QoS) across congested networks'. Checked 2026-09-24.
[^21]: Microsoft's retired Skype for Business Online page lists optimal client-to-Microsoft-edge targets of RTT below 60 ms, average packet loss below 0.5% and jitter below 3 ms. Source: [Media Quality and Network Connectivity Performance - Skype for Business Online (retired)](https://learn.microsoft.com/en-us/previous-versions/skypeforbusiness/optimizing-your-network/media-quality-and-network-connectivity-performance), section 'Network Performance requirements from a Skype for Business client to Microsoft network Edge', Optimal column. Checked 2026-09-24.
[^22]: Microsoft's retired guidance recommended testing for at least a week, sampling every 10 minutes and comparing the 90th percentile against the targets. Source: [Media Quality and Network Connectivity Performance - Skype for Business Online (retired)](https://learn.microsoft.com/en-us/previous-versions/skypeforbusiness/optimizing-your-network/media-quality-and-network-connectivity-performance), section 'Measuring network performance', high-level recommendations list. Checked 2026-09-24.
[^23]: When the connection to Webex breaks, Webex Calling endpoints switch to Survivability mode, register with the local Survivability Gateway and keep internal, PSTN and emergency calling. Source: [Site survivability for Webex Calling](https://help.webex.com/en-us/article/d68vi1/Site-Survivability-for-Webex-Calling), overview of Survivability mode and supported call types. Checked 2026-09-24.
[^24]: Cisco recommends no more than 50 ms latency between a Webex Calling Survivability Gateway and the endpoints it serves when those endpoints are at multiple locations. Source: [Site survivability for Webex Calling](https://help.webex.com/en-us/article/d68vi1/Site-Survivability-for-Webex-Calling), network requirements for Survivability Gateway (multi-location deployments). Checked 2026-09-24.
[^25]: Microsoft lists Teams one-to-one audio bandwidth per endpoint as 10/10 kbps minimum, 58/58 kbps recommended and 76/76 kbps for best performance (up/down). Source: [Prepare your organization's network for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/prepare-network), section 'Bandwidth requirements', Audio one-to-one row. Checked 2026-09-24.
[^26]: Microsoft's Teams QoS guidance recommends three to five hops from client to network edge and three hops from the ISP to the Microsoft network edge. Source: [Implement Quality of Service in Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/qos-in-teams), section 'QoS implementation checklist', guidelines list. Checked 2026-09-24.
[^27]: The Teams Network Assessment Tool runs as a single instance on Windows and is not intended for load or stress testing. Source: [Microsoft Teams Network Assessment Tool](https://www.microsoft.com/en-us/download/details.aspx?id=103017), System Requirements / Install Instructions. Checked 2026-09-24.
[^28]: The Microsoft Teams Network Assessment Tool streams packets to the nearest Teams relay edge site to measure packet loss, jitter and round-trip time, and it also checks connectivity to the required IPs and ports. Source: [Microsoft Teams Network Assessment Tool](https://www.microsoft.com/en-us/download/details.aspx?id=103017), Details / Overview description. Checked 2026-09-24.
[^29]: Microsoft's current Teams network preparation page (ms.date 2026-05-26) covers requirements, optimization tasks and bandwidth but appears to publish no numeric latency, jitter or packet loss targets (inferred). Source: [Prepare your organization's network for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/prepare-network), whole page; sections 'Network requirements', 'Network optimization', 'Bandwidth requirements'. Checked 2026-09-24.
[^30]: Teams Network Planner (Teams admin center > Planning > Network planner) calculates bandwidth requirements per site from the sites, network details and personas an admin enters. Source: [Use the Network planner for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/network-planner), introduction and section 'Create a report'. Checked 2026-09-24.
[^31]: With Microsoft's recommended personas, Network Planner automatically splits a site's users 80% office worker and 20% remote worker, and WAN and internet capacity must be entered to get bandwidth recommendations. Source: [Use the Network planner for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/network-planner), sections 'Build your plan' (note) and 'Create a report' step 2. Checked 2026-09-24.
[^32]: Using Teams Network Planner requires the Global Administrator, Teams Administrator or Teams Communications Administrator role. Source: [Use the Network planner for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/network-planner), introduction, paragraph after persona note. Checked 2026-09-24.
[^33]: Microsoft's recommended starting QoS configuration for Teams audio is client source ports 50000-50019 marked DSCP 46 (EF). Source: [Implement Quality of Service in Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/qos-in-teams), section 'Step 3. Choose initial port ranges for each media type', table 'Recommended initial port ranges'. Checked 2026-09-24.
[^34]: Microsoft says Teams QoS works as expected only when it is applied on every link between callers, so for remote users traffic can be prioritized only within the internal managed network. Source: [Implement Quality of Service in Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/qos-in-teams), section 'Step 1. Make sure your network is ready', subsection 'VPN considerations'. Checked 2026-09-24.
[^35]: Microsoft's recommended starting QoS configuration for Teams is video on 50020-50039 at DSCP 34 (AF41) and screen sharing on 50040-50059 at DSCP 18 (AF21). Source: [Implement Quality of Service in Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/qos-in-teams), section 'Step 3. Choose initial port ranges for each media type', table 'Recommended initial port ranges'. Checked 2026-09-24.
[^36]: Microsoft recommends a split-tunnel VPN so Teams traffic bypasses the VPN, noting that VPNs are usually not built for real-time media and some do not support UDP. Source: [Prepare your organization's network for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/prepare-network), section 'Network optimization', row 'Configure split-tunnel VPN'. Checked 2026-09-24.
[^37]: For Teams, Microsoft says the firewall must not change mapped NAT addresses or ports for UDP (session persistence). Source: [Prepare your organization's network for Microsoft Teams](https://learn.microsoft.com/en-us/microsoftteams/prepare-network), section 'Network optimization', row 'Maintain session persistence'. Checked 2026-09-24.
[^38]: The Webex App uses media ports 8500-8599 for audio and 8600-8699 for video on Webex Calling. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Quality of Service (QoS)/Type of Service(ToS)' and firewall port tables. Checked 2026-09-24.
[^39]: Cisco sizes Webex Calling throughput as the number of expected concurrent calls multiplied by the per-call bandwidth for the codec in use. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', bandwidth planning. Checked 2026-09-24.
[^40]: Cisco's Prepare guidance names CScan (cscan.webex.com) as the tool for assessing network quality, estimated call capacity and latency. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', CScan reference. Checked 2026-09-24.
[^41]: Webex Calling marks audio from the Webex App and from devices as DSCP 46 (Expedited Forwarding). Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Quality of Service (QoS)/Type of Service(ToS)'. Checked 2026-09-24.
[^42]: Webex Calling marks device signaling as CS3 (DSCP 24) and Webex App signaling as CS0 (DSCP 0). Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Quality of Service (QoS)/Type of Service(ToS)'. Checked 2026-09-24.
[^43]: Webex Calling marks video from the Webex App and from devices as DSCP 34 (AF41). Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Quality of Service (QoS)/Type of Service(ToS)'. Checked 2026-09-24.
[^44]: Cisco's bandwidth table uses 80 kbps per call for G.711 calls from a desk phone to the PSTN through a Local Gateway. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', bandwidth per call type table. Checked 2026-09-24.
[^45]: Filtering Webex Calling traffic by IP address alone is not supported because some Webex IP address pools are dynamic and can change at any time. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Firewall configuration'. Checked 2026-09-24.
[^46]: For Webex Calling calls over the internet, Cisco's migration Prepare guidance sets a jitter target of less than 10 ms. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements' (latency, jitter and packet loss). Checked 2026-09-24.
[^47]: For Webex Calling calls over the internet, Cisco's migration Prepare guidance sets a one-way latency target of less than 100 ms. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements' (latency, jitter and packet loss). Checked 2026-09-24.
[^48]: For Webex Calling calls over the internet, Cisco's migration Prepare guidance sets a packet loss target of 0.5% or less. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements' (latency, jitter and packet loss). Checked 2026-09-24.
[^49]: Cisco says to keep the default MTU of 1500 bytes for all IP packets carrying Webex Calling and Webex Aware services. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'MTU size for Webex Calling & Webex Aware services'. Checked 2026-09-24.
[^50]: For Webex Calling, organizations must assign enough public IP addresses to NAT pools to prevent port exhaustion. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Webex Calling traffic through proxies and firewalls' (NAT/PAT guidance). Checked 2026-09-24.
[^51]: Cisco's bandwidth table uses 70 kbps per call for OPUS calls between the Webex App or a desk phone and the Webex App. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', bandwidth per call type table. Checked 2026-09-24.
[^52]: Cisco places network readiness and requirements in the Prepare phase of a Unified CM to Webex Calling migration, which comes before the Plan phase. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), subsection 'Network readiness and requirements'; closing link to the Plan article (nw9po3o). Checked 2026-09-24.
[^53]: When a proxy is configured, only Webex Calling HTTP/HTTPS signaling goes to the proxy; SIP signaling and media do not. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Proxy support for Webex Calling'. Checked 2026-09-24.
[^54]: Cisco's Prepare guidance says to implement QoS in the on-premises environment to prioritize real-time traffic and keep it in place across switches, routers and firewalls. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', Quality of Service. Checked 2026-09-24.
[^55]: Cisco recommends turning off SIP ALG on routers and firewalls carrying Webex Calling traffic because some implementations break firewall traversal. Source: [Port Reference Information for Webex Calling](https://help.webex.com/article/b2exve/port-reference-information-for-cisco-webex-calling), section 'Firewall configuration'. Checked 2026-09-24.
[^56]: Cisco's bandwidth table allows up to 1,500 kbps per call for H.264 video calls in Webex Calling. Source: [Prepare to migrate from Unified CM to Webex Calling](https://help.webex.com/en-us/article/4yz2zd/Prepare-to-migrate-from-Unified-CM-to-Webex-Calling), section 'Network readiness and requirements', bandwidth per call type table. Checked 2026-09-24.
[^57]: Cisco's recommended network targets for good voice quality over the internet are 1-way latency below 100 milliseconds, jitter below 10 milliseconds, and packet loss of 0.5 percent or less. Source: [Transitioning from Unified CM to Webex Calling — Deployment Guide](https://www.cisco.com/c/dam/en/us/td/docs/solutions/PA/mcp/DEPLOYMENT_CALLING_Unified_CM_to_Webex_Calling.pdf), Network readiness, Latency, Jitter, and Packet Loss Targets. Checked 2026-09-04.