Member Avatar for jackspar

My project with SIP Communicator as part of the Season of Usability was to evaluate the interface of SIP Communicator, find out how SIP Communicator measured up with competing IM projects, and do an exploratory study to determine priorities for future improvements based on user requirements.

During the course of this project I work with Raphael Wimmer, my usability mentor, and Emil Ivov, creator of the SIP Communicator project, being able to work with both of these individuals helped me get a sense of the impact this work was having and would have for the future of SIP Communicator’s interface.

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jackspar.

Buy termination routes from international VoIP carriers, or sell your A-to-Z VoIP/TDM routes to a large audience.

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  • Latest Post Latest Post by xsuperstore

Dani AI

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Concurrent VoIP clients on older phones often fail for reasons unrelated to SIP itself: the handset OS and multimedia stack can treat the microphone/speaker and low-level network sockets as exclusive resources, and some clients bind fixed RTP/SIP ports or assume foreground-only operation. ’s E51 report vs ’s 5800 success points to device/firmware and client-design differences rather than a single universal bug. This kind of behaviour is both a technical and usability issue worth noting in ’s evaluation.

Quick diagnostic checklist:

  • Reproduce deliberately and record what changes: start client A, then start client B; note whether one app crashes, freezes, loses audio, or is simply deregistered. Exact symptoms matter.
  • Collect device and software details: handset model, OS/firmware version, and exact client builds. Those make the difference.
  • Test audio exclusivity: while a VoIP client is running, try playing audio or opening a voice recorder to see if the audio path is locked.
  • Test port/socket conflicts: if a client exposes local RTP/SIP port settings, try different ports or transports (UDP vs TCP) to rule out bind conflicts.
  • Try different networks: Wi‑Fi vs cellular data can trigger different client behaviors (background rules, NAT handling).
  • Practical workaround: consolidate accounts into a single client that supports multiple SIP accounts, or route calls server-side (PBX/bridge) to one active endpoint.

Developer-focused notes: mobile clients should avoid hardcoded local ports, use ephemeral ports when possible, release audio resources when backgrounded, and provide a small headless service to keep registration without claiming audio. Also document background limitations clearly for users.

Useful diagnostic data for follow-ups: device model and firmware, client versions, exact reproduction steps and any error logs or messages. With those details the community can give targeted fixes; until then, server-side routing or a multi-account client is the most reliable user-level solution.

Member Avatar for Saneha

I have faced an issue on nokia e51 the SIP device that it don't run two VoIP apps at the same time. One stop responding if you install the second one .. What you think is this a problem with devices or SIP configuration ??

Member Avatar for andy345

Hi Saneha,

I am facing the same problem did you get any solutions for this. Awaiting your reply
thanks and regards
Andy
Voip Solutions

Member Avatar for Saneha

Not yet .. Which two VoIP applications you have tested ?? I was using and fring ...

Member Avatar for xsuperstore

Hi, fring works prefect , when I'am in the offcie I redirect my extension to my nokia 5800 phone, Cell calls and internal PBX extension in the same device.

Best regards.
Max.

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