[Freeswitch-users] Frequent Call Drops in Freeswitch conference call in GSM network

[Steven Ayre]

You should be able to see in the sip trace between you and your gateway
service which side initiates the hangup. I doubt it is from the FreeSWITCH
side. One thing that might be worth checking is whether there are SIP
session timers or RTP timers in use that might be timing out.

On 17 April 2017 at 08:26, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi,
>
> I involved in a research project where we have developed a mobile based
> training system. Our server is Freeswitch based. The server on receiving
> request from an android app establishs conference call amongst 21
> participants over the cellular network by calling on their mobile numbers.
> To achieve this, a VOIP-GSm gateway service is used offered by Doorvaani.
> The participants locations are rural areas of India. We need to maintain
> the conference call for atleast 1 hour as there is some kind of training
> program.
>
> We have tested our training platform by conducting 10 such conference
> calls. However, one persistent problem that we are facing is frequent call
> drops. In between of an on-going conference, some of the participants faces
> call drops, to which we have adopted a workaround of redialing them.
>
> One of the potential cause is the poor signal state in some areas, but
> there are many people who have decent signal strength and still face call
> drops.
>
> We are not sure how can we investigate the matter as neither the VOIP-GSM
> logs are available for us nor can we setup conference call with more than
> 5-6 participants using existing service providers.
>
> Is this a problem from Freeswitch, can I debug? or should we setup our own
> VOIP-GSM gateway as currently we are using web service from a company we
> can not physically access the logs. I would be happy to get suggestions on
> how to deal with this.
>
> Regards
> Deepika
> https://www.iiitd.edu.in/~deepikay/
>
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