DEBUG rtp_engine.c: Copying tx payload mapping 101 (0x7f4da807eab8) from 0x7f4e04ce6fc0 to 0x7f4da80325a8 DEBUG rtp_engine.c: Copying tx payload mapping 97 (0x7f4da8099d78) from 0x7f4e04ce6fc0 to 0x7f4da80325a8 DEBUG rtp_engine.c: Copying tx payload mapping 8 (0x5602af20f7a8) from 0x7f4e04ce6fc0 to 0x7f4da80325a8 DEBUG rtp_engine.c: Copying tx payload mapping 3 (0x5602af20f618) from 0x7f4e04ce6fc0 to 0x7f4da80325a8 DEBUG rtp_engine.c: Copying tx payload mapping 0 (0x5602af20f5c8) from 0x7f4e04ce6fc0 to 0x7f4da80325a8 DEBUG rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e04ce6fc0 DEBUG rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f4e04ce6fc0 DEBUG rtp_engine.c: Setting tx payload type 3 based on m type on 0x7f4e04ce6fc0 DEBUG res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4da80323d0' DEBUG res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4da80323d0' DEBUG res_pjsip_session.c: Applying negotiated SDP media stream 'audio' using audio SDP handler DEBUG channel.c: Channel PJSIP/6001-00000002 setting read format path: ulaw -> ulaw DEBUG channel.c: Channel PJSIP/6001-00000002 setting write format path: ulaw -> ulaw DEBUG res_pjsip/pjsip_resolver.c: Target '192.168.0.7' is an IP address, skipping resolution DEBUG res_pjsip/pjsip_resolver.c: Transport type for target '192.168.0.7' is 'UDP transport' DEBUG res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '192.168.0.7' DEBUG res_pjsip_session.c: Applied negotiated SDP media stream 'audio' using audio SDP handler DEBUG channel.c: Channel PJSIP/6002-00000003 setting write format path: ulaw -> ulaw DEBUG channel.c: Channel PJSIP/6002-00000003 setting read format path: ulaw -> ulaw DEBUG rtp_engine.c: Copying tx payload mapping 101 (0x7f4da803f928) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Copying tx payload mapping 3 (0x5602af20f618) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Copying tx payload mapping 0 (0x5602af20f5c8) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Setting tx payload type 3 based on m type on 0x7f4e04ce6df0 DEBUG rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e04ce6df0 DEBUG res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4da807ecb0'
DEBUG res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4da807ecb0'
#ZOIPER PJSIP FULL#
Here is the full log after the call goes through: DEBUG res_pjsip_session.c: Applying negotiated SDP media stream 'audio' using audio SDP handler
192.168.0.2 in both clients as host then everything is perfect. However, when I put both clients behind the NAT use local ip i.e. Problem I am having is when one client calls the other, the call goes through, but no audio is present in either client. I have configured my router to forward 5060/UDP and ports 10000-20000/UDP as well. Another Client is an iPhone running on 4G network. My Asterisk and one of the clients using Zoiper Softphone are behind NAT.
#ZOIPER PJSIP PC#
I have just installed and configured Asterisk 17 in a desktop PC running Ubuntu 18.4
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