Flowroute takes a unique approach to deliver call audio via Direct Media . Direct Media allows us to offer on-demand capacity rather than limiting the number of simultaneous calls that can be made on a Flowroute account. We can offer this unlimited capacity by eliminating the need to proxy call audio between your system and the Public Switched Telephone Network (PSTN) . Direct Media facilitates the audio (RTP media) on your calls being delivered directly from your equipment to a peering network’s media gateway on the PSTN. Our peering networks’ media gateways offer massive amounts of bandwidth; using Direct Media for your calls eliminates a potential point of failure or latency in your audio stream. Direct Media allows us to offer competitive rates and great domestic and international connections for all of our customers, regardless of geographic location. IMPORTANT: Keep Direct Media in mind while configuring your SIP device …

… east-va.sip.flowroute.com 34.226.36.32/28 ASIA PACIFIC AP-East-HK ap-east-hk.sip.flowroute.com 16.163.86.112/30 AP-Southeast-SIN ap-southeast-sin.sip.flowroute.com 3.0.5.12/30 EMEA EU-West-LDN eu-west-ldn.sip.flowroute.com 3.8.37.20/30 EU-Central-FRA eu-central-fra.sip.flowroute.com 3.71.103.56/30 LATIN AMERICA SA-East-SP sa-east-sp.sip.flowroute.com 18.228.70.48/30 RTP media (call audio) To reduce latency, Flowroute uses Direct Media Delivery. To receive Direct Media, allow UDP packets from any source IP address with a destination port within your system's RTP public media port range. RTP media port range varies by phone system. Your system's RTP media port range will be configured locally on your order and detailed in your system documentation.

To ensure the best audio quality on your Flowroute calls, implement a network Quality of Service (QoS) strategy that prioritizes Flowroute's Direct Media packets. QoS strategies Give priority to UDP traffic (RTP audio) on your network over all ports within your phone system's RTP media port range Give your phone system a higher priority (by local IP address) for all network traffic. Either of these methods will optimize audio quality on Flowroute calls by helping to prevent jitter, choppiness, echoes, audio delays, pops, clicks, and intermittent garbled/robotic voices which could be introduced by local network delays if these QoS policies aren't in place. NOTE: RTP media port range varies by phone system. Your system's RTP media port range will be configured locally on your system and/or detailed in your system documentation.

Please refer to Technical Specifications for a complete service guide. Codecs Supported (SDP media attribute): G.711 Ulaw (0) G.711 Alaw (8) DTMF tones (101) G.729a (18) Flowroute takes a unique approach to deliver call audio: via Direct Media . RTP media on your calls is delivered directly from your equipment to a peering network’s media gateway on the PSTN. Read more about Direct Media in this KB article .

… East-NJ.sip.flowroute.com CIDR: 147.75.65.192/28  (Alternative) US-East-VA Edge Proxy Information: SRV: US-East-VA.sip.flowroute.com CIDR: 34.226.36.32/28 NOTE: In-flight calls will be unaffected. The decommissioning of the New Jersey Edge Proxy will be permanent and will not be restored at a later date. For US customers, OR and VA Edge proxies will remain redundant alternatives for each other. Our unique Direct Media Delivery technology allows this change with NO impact on audio quality, including when your PBX is located on the East coast and connecting to the West coast Edge Proxy. ACTION REQUIRED: Edge proxies found on the Flowroute Interconnection page will need to be added to your current firewall whitelist. You will also need to replace the existing US-East-NJ EP with another EP. We recommend connecting to Virginia Edge Proxy (US-East …

… a higher priority than other network traffic, such as checking your email. Users will typically never notice a few millisecond delays in downloading data when checking email – whereas a 400-millisecond delay in call audio packets (RTP) can be significant enough to cause choppy or jittery audio. Using QoS priorities allows you to prioritize the traffic that you would notice a small delay in – i.e. call audio (RTP) – above the traffic you wouldn't notice a small delay in – checking your email, browsing the internet, etc. CoS Additionally, your packets flagged for high QoS priority may be honored and expedited on your ISP over the public internet (or they may offer CoS – "Class of Service"). Contact your ISP for information about the honoring of QoS and/or CoS on their services. We recommend that you configure QoS on your network to prioritize all packets delivered via Flowroute Direct Media.

… Using a single number for both text and voice calls is recommended. Use one recognizable domain name Each campaign should be associated with a single web domain. A full domain is required; a URL shortener cannot be used to deliver custom links. Use natural language You should use natural language in your messages, which means that you do not use non-standard spellings. For example, “H! h0w ar3__you do1ng?” is a non-standard spelling. Direct consent You should collect the consumer consent yourself, and not use consent acquired from a third party. The consumer is expecting a relationship with the business they interacted with. Set expectations on frequency You should set the proper expectation with the consumer for informational or promotional messages. If you are sending 5 texts a month, then disclosing “5/msg a month” on the first interaction is a positive user experience. SMS/MMS Requirements and …

… s) from which a call was diverted and the diversion reason(s). Session Timers Defined in RFC 4028. Allow for establishing a call expiration interval to avoid runaway calls if a connection is lost. Optional. ISUP-OLI & JIP Where available, can provide additional calling party info. Custom Tags Allow logging up to 32 character strings in your call CDRs via an X-Tag header in either initial INVITE or answering 200 OK. Codecs Supported (SDP media attribute) G.711 Ulaw (0) G.711 Alaw (8) DTMF tones (101) G.729a (18) Recommended fax transmission guidelines Session negotiation : Switch to T.38 is dependent on end-user system reINVITE during an established G.711 audio codec session. Fallback : End-user system should support fallback to G.711 ulaw pass-through in the event of T.38 negotiation failure. Maximum transmission rate : 9600 baud. Fax Error Correction Mode (ECM) : Not supported. T …