Hi, Try 200. See http://www.ietf.org/rfc/rfc3261.txt page 12. Arie > Date: Wed, 27 Oct 2010 12:46:13 -0400 > From: abhat002@xxxxxxxxxxx > To: pjsip at lists.pjsip.org > Subject: [Fwd: No sound is heard at local/remote end!] > > Hello, > > In continuation with the below mail regarding my problem of no sound heard > at both ends, I have the following issues: > > (1) I have connected both the machines in a LAN environment without any > firewalls and able to ping each other. My call initiator machine is > 192.168.0.8 and call recvr machine is 192.168.0.16. > I am performing: ./pjsua sip:192.168.0.16 from call initiator machine. Is > this correct? > > (2) I have gone thorough the documentation for solving audio problems and > tested all of them. Everything is fine as regards to local record/playback > at both the machines. But after initiating ./pjsua sip:192.168.0.16 the > ports at the initiator machine with cl command look as follows: > Conference ports: > Port ##00[16KHz/20ms/1] HDA Intel: AD198X Analog (hw:0,0) (44KHz) > transmitting to: > Port ##01[16KHz/20ms/1] ringback transmitting to: > Port ##02[16KHz/20ms/1] ring transmitting to: > > But according to the documentation, there should be one entry in the > conference bridge with the destination sip connection. So, I am also not > able to perform cc for connecting the call to the devices. Why this > problem is happening? > > (3) Even though, a prompt appears at the rcvr end for accepting the call > and after pressing a to accept then it asks for code (100-699) and on > entering 100 nothing happens and a beep sound continues at the rcvr end. > What does this code (100-699) imply and how should I connect? By entering > 100 should I be able to connect? > Even though the call prompt is received means that network connection is > OK, then why the call is not connected to the conference bridge and no > sound is heard? > > Your help is highly appreciated, > Thanks, > Abhishek > > > > > ---------------------------- Original Message ---------------------------- > Subject: No sound is heard at local/remote end! > From: "Abhishek Bhattacharya" <abhat002@xxxxxxxxxxx> > Date: Tue, October 26, 2010 4:32 pm > To: pjsip at lists.pjsip.org > -------------------------------------------------------------------------- > > Hello, > > I am using the pjsua application to connect another machine in a LAN for a > peer-to-peer call. I have checked all the playback and recording > functionalities locally at both ends and all are perfect. > When I call another machine with IP address, the call gets connected and > prompt is received at the rcvr end for answering the call. Then it asks to > send a code (100-699) and after sending no sound is heard either ways. > I would like to know what does this code (100-699) imply! > Also, in the whole process only beep sound is heard continuously. > I am printing the console messages at both ends. > > At call initiator end: > > ****************************************************************************** > [user at localhost bin]$ ./pjsua-i686-pc-linux-gnu sip:192.168.0.16 > 15:43:40.044 os_core_unix.c pjlib 1.0.3 for POSIX initialized > 15:43:40.045 sip_endpoint.c Creating endpoint instance... > 15:43:40.045 pjlib select() I/O Queue created (0x915b1d0) > 15:43:40.045 sip_endpoint.c Module "mod-msg-print" registered > 15:43:40.045 sip_transport. Transport manager created. > 15:43:40.045 sip_endpoint.c Module "mod-pjsua-log" registered > 15:43:40.045 sip_endpoint.c Module "mod-tsx-layer" registered > 15:43:40.045 sip_endpoint.c Module "mod-stateful-util" registered > 15:43:40.045 sip_endpoint.c Module "mod-ua" registered > 15:43:40.045 sip_endpoint.c Module "mod-100rel" registered > 15:43:40.045 sip_endpoint.c Module "mod-pjsua" registered > 15:43:40.045 sip_endpoint.c Module "mod-invite" registered > 15:43:40.084 pasound.c PortAudio sound library initialized, status=0 > 15:43:40.084 pasound.c PortAudio host api count=2 > 15:43:40.084 pasound.c Sound device count=10 > 15:43:40.084 pjlib select() I/O Queue created (0x917f974) > 15:43:40.084 sip_endpoint.c Module "mod-evsub" registered > 15:43:40.084 sip_endpoint.c Module "mod-presence" registered > 15:43:40.084 sip_endpoint.c Module "mod-refer" registered > 15:43:40.084 sip_endpoint.c Module "mod-pjsua-pres" registered > 15:43:40.084 sip_endpoint.c Module "mod-pjsua-im" registered > 15:43:40.084 sip_endpoint.c Module "mod-pjsua-options" registered > 15:43:40.084 pjsua_core.c 1 SIP worker threads created > 15:43:40.084 pjsua_core.c pjsua version 1.0.3 for i686-pc-linux-gnu > initialized > 15:43:40.084 sip_endpoint.c Module "mod-default-handler" registered > 15:43:40.085 pjsua_core.c SIP UDP socket reachable at 192.168.0.8:5060 > 15:43:40.085 udp0x9190020 SIP UDP transport started, published address > is 192.168.0.8:5060 > 15:43:40.085 pjsua_acc.c Account <sip:192.168.0.8:5060> added with id 0 > 15:43:40.085 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.0.8:5060 > 15:43:40.085 pjsua_acc.c Account <sip:192.168.0.8:5060;transport=TCP> > added with id 1 > 15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4000 > 15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4001 > 15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4002 > 15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4003 > 15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4004 > 15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4005 > 15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4006 > 15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4007 > 15:43:40.085 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @16000 Hz > 15:43:40.088 pjsua_media.c ..failed: Invalid sample rate > 15:43:40.088 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @44100 Hz > 15:43:40.128 os_core_unix.c Info: possibly re-registering existing thread > 15:43:40.217 ec0x917ee18 AEC created, clock_rate=44100, channel=1, > samples per frame=882, tail length=200 ms, latency=88969 ms > 15:43:40.217 pjsua_call.c Making call with acc #1 to sip:192.168.0.16 > 15:43:40.228 pjsua_media.c Media index 0 selected for call 0 > 15:43:40.228 pjsua_core.c TX 1020 bytes Request msg INVITE/cseq=431 > (tdta0x9ad4d40) to UDP 192.168.0.16:5060: > INVITE sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16 > Contact: <sip:192.168.0.8:5060> > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu > Content-Type: application/sdp > Content-Length: 456 > > v=0 > o=- 3497111020 3497111020 IN IP4 192.168.0.8 > s=pjmedia > c=IN IP4 192.168.0.8 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 > a=rtcp:4001 IN IP4 192.168.0.8 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:117 iLBC/8000 > a=fmtp:117 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > 15:43:40.228 pjsua_app.c Call 0 state changed to CALLING > >>>> > Account list: > [ 0] <sip:192.168.0.8:5060>: does not register > Online status: Online > *[ 1] <sip:192.168.0.8:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > [ 1] <?> sip:192.168.0.16 > > +=============================================================================+ > | Call Commands: | Buddy, IM & Presence: | Account: > | > | | | > | > | m Make new call | +b Add new buddy .| +a Add new > accnt | > | M Make multiple calls | -b Delete buddy | -a Delete > accnt. | > | a Answer call | i Send IM | !a Modify > accnt. | > | h Hangup call (ha=all) | s Subscribe presence | rr > (Re-)register | > | H Hold call | u Unsubscribe presence | ru Unregister > | > | v re-inVite (release hold) | t ToGgle Online status | > Cycle next > ac.| > | U send UPDATE | T Set online status | < Cycle prev > ac.| > | ],[ Select next/prev call > +--------------------------+-------------------+ > | x Xfer call | Media Commands: | Status & > Config: | > | X Xfer with Replaces | | > | > | # Send RFC 2833 DTMF | cl List ports | d Dump > status | > | * Send DTMF with INFO | cc Connect port | dd Dump > detailed | > | dq Dump curr. call quality | cd Disconnect port | dc Dump > config | > | | V Adjust audio Volume | f Save > config | > | S Send arbitrary REQUEST | Cp Codec priorities | f Save > config | > +------------------------------+--------------------------+-------------------+ > | q QUIT sleep MS echo [0|1|txt] n: detect NAT type > | > +=============================================================================+ > You have 1 active call > Current call id=0 to sip:192.168.0.16 [CALLING] > >>> 15:43:40.241 pjsua_core.c RX 317 bytes Response msg > 100/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16> > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:43:45.229 sound_port.c EC suspended because of inactivity > 15:43:51.065 pjsua_core.c RX 317 bytes Response msg > 100/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16> > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:44:11.801 pjsua_core.c RX 359 bytes Response msg > 603/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060: > SIP/2.0 603 Decline > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:44:11.801 pjsua_core.c TX 355 bytes Request msg ACK/cseq=431 > (tdta0x9ad74f8) to UDP 192.168.0.16:5060: > ACK sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 ACK > Content-Length: 0 > > > --end msg-- > 15:44:11.801 pjsua_app.c Call 0 is DISCONNECTED [reason=603 (Decline)] > 15:44:11.801 pjsua_app.c > [DISCONNCTD] To: sip:192.168.0.16 > Call time: 00h:00m:00s, 1st res in 31584 ms, conn in 0ms > SRTP status: Not active Crypto-suite: (null) > 15:44:13.305 pjsua_core.c RX 359 bytes Response msg > 603/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060: > SIP/2.0 603 Decline > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:44:13.305 pjsua_core.c TX 355 bytes Request msg ACK/cseq=431 > (tdta0x9ad74f8) to UDP 192.168.0.16:5060: > ACK sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 ACK > Content-Length: 0 > > > --end msg-- > q > 15:44:18.332 pjsua_media.c Closing (null) sound playback device and > (null) sound capture device > 15:44:19.638 pasound.c PortAudio sound library shutting down.. > 15:44:19.638 pjsua_core.c Shutting down... > 15:44:20.645 pjsua_core.c Destroying... > 15:44:20.645 sip_transactio Stopping transaction layer module > 15:44:20.646 sip_endpoint.c Module "mod-default-handler" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-pjsua-options" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-pjsua-im" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-pjsua-pres" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-pjsua" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-stateful-util" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-refer" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-presence" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-evsub" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-invite" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-100rel" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-ua" unregistered > 15:44:20.646 sip_transactio Transaction layer module destroyed > 15:44:20.646 sip_endpoint.c Module "mod-tsx-layer" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-msg-print" unregistered > 15:44:20.646 sip_endpoint.c Module "mod-pjsua-log" unregistered > 15:44:20.647 tcplis:5060 SIP TCP listener destroyed > 15:44:20.647 sip_endpoint.c Endpoint 0x9153324 destroyed > 15:44:20.647 pjsua_core.c PJSUA destroyed... > [user at localhost bin]$ > *************************************************************************** > > At the call receiver end: > > **************************************************************************** > [user at localhost bin]$ ./pjsua-i686-pc-linux-gnu > 15:48:56.123 os_core_unix.c pjlib 1.0.3 for POSIX initialized > 15:48:56.123 sip_endpoint.c Creating endpoint instance... > 15:48:56.124 pjlib select() I/O Queue created (0x86fd1d0) > 15:48:56.124 sip_endpoint.c Module "mod-msg-print" registered > 15:48:56.124 sip_transport. Transport manager created. > 15:48:56.124 sip_endpoint.c Module "mod-pjsua-log" registered > 15:48:56.124 sip_endpoint.c Module "mod-tsx-layer" registered > 15:48:56.124 sip_endpoint.c Module "mod-stateful-util" registered > 15:48:56.124 sip_endpoint.c Module "mod-ua" registered > 15:48:56.124 sip_endpoint.c Module "mod-100rel" registered > 15:48:56.124 sip_endpoint.c Module "mod-pjsua" registered > 15:48:56.124 sip_endpoint.c Module "mod-invite" registered > 15:48:56.164 pasound.c PortAudio sound library initialized, status=0 > 15:48:56.164 pasound.c PortAudio host api count=2 > 15:48:56.164 pasound.c Sound device count=10 > 15:48:56.164 pjlib select() I/O Queue created (0x872192c) > 15:48:56.164 sip_endpoint.c Module "mod-evsub" registered > 15:48:56.164 sip_endpoint.c Module "mod-presence" registered > 15:48:56.164 sip_endpoint.c Module "mod-refer" registered > 15:48:56.164 sip_endpoint.c Module "mod-pjsua-pres" registered > 15:48:56.164 sip_endpoint.c Module "mod-pjsua-im" registered > 15:48:56.164 sip_endpoint.c Module "mod-pjsua-options" registered > 15:48:56.164 pjsua_core.c 1 SIP worker threads created > 15:48:56.164 pjsua_core.c pjsua version 1.0.3 for i686-pc-linux-gnu > initialized > 15:48:56.164 sip_endpoint.c Module "mod-default-handler" registered > 15:48:56.164 pjsua_core.c SIP UDP socket reachable at 192.168.0.16:5060 > 15:48:56.164 udp0x87320d0 SIP UDP transport started, published address > is 192.168.0.16:5060 > 15:48:56.165 pjsua_acc.c Account <sip:192.168.0.16:5060> added with id 0 > 15:48:56.165 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.0.16:5060 > 15:48:56.165 pjsua_acc.c Account > <sip:192.168.0.16:5060;transport=TCP> added with id 1 > 15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4000 > 15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4001 > 15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4002 > 15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4003 > 15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4004 > 15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4005 > 15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4006 > 15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4007 > >>>> > Account list: > [ 0] <sip:192.168.0.16:5060>: does not register > Online status: Online > *[ 1] <sip:192.168.0.16:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > -none- > > +=============================================================================+ > | Call Commands: | Buddy, IM & Presence: | Account: > | > | | | > | > | m Make new call | +b Add new buddy .| +a Add new > accnt | > | M Make multiple calls | -b Delete buddy | -a Delete > accnt. | > | a Answer call | i Send IM | !a Modify > accnt. | > | h Hangup call (ha=all) | s Subscribe presence | rr > (Re-)register | > | H Hold call | u Unsubscribe presence | ru Unregister > | > | v re-inVite (release hold) | t ToGgle Online status | > Cycle next > ac.| > | U send UPDATE | T Set online status | < Cycle prev > ac.| > | ],[ Select next/prev call > +--------------------------+-------------------+ > | x Xfer call | Media Commands: | Status & > Config: | > | X Xfer with Replaces | | > | > | # Send RFC 2833 DTMF | cl List ports | d Dump > status | > | * Send DTMF with INFO | cc Connect port | dd Dump > detailed | > | dq Dump curr. call quality | cd Disconnect port | dc Dump > config | > | | V Adjust audio Volume | f Save > config | > | S Send arbitrary REQUEST | Cp Codec priorities | f Save > config | > +------------------------------+--------------------------+-------------------+ > | q QUIT sleep MS echo [0|1|txt] n: detect NAT type > | > +=============================================================================+ > You have 0 active call > >>> 15:49:29.163 pjsua_core.c RX 1020 bytes Request msg > INVITE/cseq=431 (rdata0x8732544) from UDP 192.168.0.8:5060: > INVITE sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16 > Contact: <sip:192.168.0.8:5060> > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu > Content-Type: application/sdp > Content-Length: 456 > > v=0 > o=- 3497111020 3497111020 IN IP4 192.168.0.8 > s=pjmedia > c=IN IP4 192.168.0.8 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 > a=rtcp:4001 IN IP4 192.168.0.8 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:117 iLBC/8000 > a=fmtp:117 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > 15:49:29.173 pjsua_media.c Media index 0 selected for call 0 > 15:49:29.173 pjsua_core.c TX 317 bytes Response msg > 100/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16> > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:49:29.173 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @16000 Hz > 15:49:29.176 pjsua_media.c ..failed: Invalid sample rate > 15:49:29.176 pjsua_media.c pjsua_set_snd_dev(): attempting to open > devices @44100 Hz > 15:49:29.208 os_core_unix.c Info: possibly re-registering existing thread > 15:49:29.296 ec0x8720d98 AEC created, clock_rate=44100, channel=1, > samples per frame=882, tail length=200 ms, latency=88969 ms > 15:49:29.296 conference.c Port 2 (ring) transmitting to port 0 (HDA > Intel: AD198x Analog (hw:0,0) (44KHz)) > 15:49:29.296 pjsua_app.c Incoming call for account 0! > From: <sip:192.168.0.8> > To: <sip:192.168.0.16> > Press a to answer or h to reject call > a > Answer with code (100-699) (empty to cancel): 100 > 15:49:39.999 pjsua_core.c TX 317 bytes Response msg > 100/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16> > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > >>> q > 15:50:00.736 pjsua_core.c TX 359 bytes Response msg > 603/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060: > SIP/2.0 603 Decline > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:50:00.736 pjsua_app.c Call 0 is DISCONNECTED [reason=603 (Decline)] > 15:50:00.736 pjsua_app.c > [DISCONNCTD] To: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > Call time: 00h:00m:00s, 1st res in 10836 ms, conn in 0ms > SRTP status: Not active Crypto-suite: (null) > 15:50:00.736 pjsua_media.c Closing (null) sound playback device and > (null) sound capture device > 15:50:02.239 pasound.c PortAudio sound library shutting down.. > 15:50:02.240 pjsua_core.c Shutting down... > 15:50:02.240 pjsua_core.c TX 359 bytes Response msg > 603/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060: > SIP/2.0 603 Decline > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > CSeq: 431 INVITE > Content-Length: 0 > > > --end msg-- > 15:50:02.240 pjsua_core.c RX 355 bytes Request msg ACK/cseq=431 > (rdata0x8732544) from UDP 192.168.0.8:5060: > ACK sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 ACK > Content-Length: 0 > > > --end msg-- > 15:50:02.240 pjsua_core.c RX 355 bytes Request msg ACK/cseq=431 > (rdata0x8732544) from UDP 192.168.0.8:5060: > ACK sip:192.168.0.16 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6 > Max-Forwards: 70 > From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146 > To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19 > Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462 > CSeq: 431 ACK > Content-Length: 0 > > > --end msg-- > 15:50:03.248 pjsua_core.c Destroying... > 15:50:03.248 sip_transactio Stopping transaction layer module > 15:50:03.248 sip_endpoint.c Module "mod-default-handler" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-pjsua-options" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-pjsua-im" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-pjsua-pres" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-pjsua" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-stateful-util" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-refer" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-presence" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-evsub" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-invite" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-100rel" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-ua" unregistered > 15:50:03.248 sip_transactio Transaction layer module destroyed > 15:50:03.248 sip_endpoint.c Module "mod-tsx-layer" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-msg-print" unregistered > 15:50:03.248 sip_endpoint.c Module "mod-pjsua-log" unregistered > 15:50:03.249 tcplis:5060 SIP TCP listener destroyed > 15:50:03.249 sip_endpoint.c Endpoint 0x86f5324 destroyed > 15:50:03.249 pjsua_core.c PJSUA destroyed... > [user at localhost bin]$ > **************************************************************************** > > Any help will be highly appreciated! > > Thanks and Regards, > Abhishek Bhattacharya > > > Regards, > Abhishek Bhattacharya > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip 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