On 13/8/25 19:48, Karthikeyan via Users wrote:
Team, We have implemented OpenSIPS Server 3.5 in Ubuntu 24,04,We were able to configure the entire setup through the following YouTube videoshttps://www.youtube.com/watch?v=o5BW0r1RKFohttps://www.youtube.com/watch?v=QJuKMq3W86U&t=13s <https://www.youtube.com/watch?v=QJuKMq3W86U&t=13s> https://www.youtube.com/watch?v=33xtIuCyWhw&t=149s <https://www.youtube.com/watch?v=33xtIuCyWhw&t=149s>https://www.youtube.com/watch?v=1KL2kMkdWcohttps://www.youtube.com/watch?v=oweTSW61cH8&list=PLJ_HEpr29ISM3Mj9PhJBlKr9GAxSeohAj&index=15 <https://www.youtube.com/watch?v=oweTSW61cH8&list=PLJ_HEpr29ISM3Mj9PhJBlKr9GAxSeohAj&index=15>We are facing following issues 1. In some phones when we make SIP call there is no voice but in other phones, we can hear the voice clearly. 2. When we make SIP call with in the same WIFI network then there is no voice. 3. SIP call is not getting disconnected even when the call ends.We have attached the OpenSIPS config we have implemented for your reference.Please help us to fix these issues. Regards Karthik
The SIP voice and SIP command channels are completely separate IP systems. One is the Session Initiation Protocol (SIP) that establishes and controls a call and one is RTP which is Real Time Transport protocol (RTP) that carries the audio. For voice to work between devices the RTP protocol must be in the same LAN network, on public IP addresses, or have a proxy to communicate.
Voice not working says the SIP is communicating correctly but RTP is not working in or across networks.
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