A computer-implemented method includes generating at least one call detail record for each call initiated on a communication network, aggregating the call detail records generated over a predetermined time period to provide an aggregated call detail record, and calculating a call set up failure rate for the calls from the aggregated call detail record, wherein the call set up failure rate is a measure of the amount of calls that failed prior to ringing at a called party's device. The computer-implemented method further includes adjusting parameters of the communication network when the call set up failure rate is above a threshold.

Systems, methods, and computer program products are provided for improving establishment and streaming of audio communication among mobile computing devices and for handling dropping or adding of users during an audio conversation. For example, a method comprises: establishing, on a mobile application, an audio conversation among at least a first user and a second user; streaming the audio conversation to a third user who accesses the mobile application on a mobile device; transmitting to the mobile device for visual display, during the audio conversation, on the user interface of the mobile application on the mobile device, a visual representation of the first user and the second user; determining the second user drops out of the audio conversation; and adding a new user to the audio conversation.

A method for emergency communication includes receiving, by a wireless device, a user instruction or an automatically triggered event for reporting an emergency to an emergency monitoring server; and in response to the user instruction or the automatically triggered event, transmitting, by the wireless device, a data-only report to the emergency monitoring server.

Methods and architectures for improving operation of 9-1-1 and other emergency services networks. Enriched data (e.g., text messages, videos, phase II location information, etc.) may be transmitted to a PSAP via a secondary communication channel (i.e., data), wherein the primary communication channel is a voice telephony trunk. Disclosed embodiments may increase the speed with which accurate location information is made available to improve call routing accuracy and decrease dispatch time. Some embodiments may also enable continued operation of legacy emergency services systems during and after transition to next generation systems.

Systems, methods, and computer program products are provided for improving establishment and broadcasting of communication, including improving speaker-listener engagement using audio conversation control. For example, a method comprises determining a first user accesses a mobile application on a first mobile device of the first user; determining a second user accesses the mobile application on a second mobile device of the second user; initiating an audio conversation between the first mobile device of the first user and the second mobile device of the second user; and receiving an instruction from the first mobile device to enable a third user who accesses the mobile application on a third mobile device of the third user to listen to the audio conversation, wherein the instructions prevents a fourth user who accesses the mobile application on a fourth mobile device of the fourth user to listen to the audio conversation.

Described herein are techniques, devices, and systems for providing an optimal voice experience over varying radio frequency (RF) conditions while using EVS audio codecs. A user equipment (UE) may adaptively transition between using a music-capable EVS codec (e.g., EVS-FB) as a default audio codec that provides a first audio bandwidth and a different EVS audio codec that provides a second audio bandwidth that is less than the first audio bandwidth. The transition to the different EVS audio codec may occur in response to determining a value indicative of a RF condition associated with a serving base station is less than a threshold value, which allows for providing preserving at least a minimal level of voice quality in degraded RF conditions.

Described herein are techniques, devices, and systems for providing an optimal voice experience over varying radio frequency (RF) conditions while using EVS audio codecs. A user equipment (UE) may adaptively transition between using a music-capable EVS codec (e.g., EVS-FB) as a default audio codec that provides a first audio bandwidth and a different EVS audio codec that provides a second audio bandwidth that is less than the first audio bandwidth. The transition to the different EVS audio codec may occur in response to determining a value indicative of a RF condition associated with a serving base station is less than a threshold value, which allows for providing preserving at least a minimal level of voice quality in degraded RF conditions.

A computer-implemented method includes generating at least one call detail record for each call initiated on a communication network, aggregating the call detail records generated over a predetermined time period to provide an aggregated call detail record, and calculating a call set up failure rate for the calls from the aggregated call detail record, wherein the call set up failure rate is a measure of the amount of calls that failed prior to ringing at a called party's device. The computer-implemented method further includes adjusting parameters of the communication network when the call set up failure rate is above a threshold.

A method performed by a network node, for handling an outgoing media call of a user associated with multiple User Equipment, UEs, in a communications network is provided. The network node receives (402) a request for the outgoing media call to be set up from a first UE out of the multiple UEs. The request is obtained as a voice instruction 5 from the user via a Digital Assistant, DA, device. The voice instruction comprises a personalized name associated with the first UE. The network node further identifies (403) among the multiple UEs, the first UE and its associated IP Multimedia Subsystem, IMS, identifier based on the personalized name received in the voice instruction. The network node further sends (404), to an IMS node, the identified IMS identifier of the first UE and a 10 request to whether or not the first UE is registered in the IMS node and supports media When the first UE is registered in the IMS node and supports the media of the outgoing media call, the network node initiates (405) the outgoing media call from the first UE. When the first UE is not registered in the IMS node and/or does not support the media of the outgoing media call, the network node decides (406) how to 15 proceed with the outgoing media call.

It is provided a method, comprising monitoring if an invitation to an emergency session of an anonymous user is received from an internet protocol address; checking, if the invitation is received, if an identity of a user is correlated to the internet protocol address; replacing, if the internet protocol address is correlated to the identity of the user, the identification of the anonymous user in the invitation by the identity of the user; forwarding the invitation comprising the identity of the user.