[pjsip] Different codec priorities in 1xx and 200

Alexei Kuznetsov eofster at gmail.com
Tue Apr 21 09:34:47 EDT 2009


Hi,

I've noticed a one-way audio problem with one of the SIP providers.  
It's definitely a media issue, not a network issue. I make a call,  
another party answers and hears me, but I can't her another party.  
When he or she starts speaking, the output on my side says

"strm0x98fd74  Bad RTP pt 8 (expecting 117)"

I've noticed that there are different codec priorities in 183 and in  
200 replies from that provider. When it replies with 183 with iLBC as  
the most preferred codec, pjsua-lib sais

pjsua_media.c  Media updates, stream #0: iLBC (sendrecv)

Later, when provider replies with 200 and PCMA as the most preferred  
codec, there are no "Media updates" in the pjsua-lib log output and  
there is "SDP negotiation done, message body is ignored" instead. And  
error messages start appearing when another party speaks.

Is such behaviour of that SIP provider appropriate? Should pjsua-lib  
adapt to the codec changes between 1xx and 2xx?

Alexei



17:20:38.866   pjsua_core.c  TX 1318 bytes Request msg INVITE/ 
cseq=31721 (tdta0x85fa00) to UDP 217.73.112.14:5060:
INVITE sip:012345678901 at sip.pctel.ru SIP/2.0
Via: SIP/2.0/UDP  
91.78.58.52:5060;rport;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb
Max-Forwards: 70
From: sip:johnsmith at sip.pctel.ru;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3
To: sip:012345678901 at sip.pctel.ru
Contact: <sip:johnsmith at 91.78.58.52:5060>
Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e
CSeq: 31721 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,  
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: PJSUA v1.0.2/i386-apple-darwin9.6.0
Authorization: Digest username="johnsmith", realm="sip.pctel.ru",  
nonce="49edc952a4cf810a00e6bae7252c2707601bf829", uri="sip:012345678901 at sip.pctel.ru 
", response="c8a54e3a5f7d1d4a9e15549f46f527e3",  
cnonce="hsrQ4p0kagPMvuwq6OMEMpnGuEIMX9re", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   456

v=0
o=- 3449308838 3449308838 IN IP4 91.78.58.52
s=pjmedia
c=IN IP4 91.78.58.52
t=0 0
a=X-nat:8
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101
a=rtcp:4001 IN IP4 91.78.58.52
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--
  17:20:38.866    pjsua_app.c  Call 0 state changed to CALLING
  17:20:38.906   pjsua_core.c  RX 309 bytes Response msg 100/INVITE/ 
cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060:
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP  
91.78.58.52 
:5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb
From: sip:johnsmith at sip.pctel.ru;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3
To: sip:012345678901 at sip.pctel.ru
Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e
CSeq: 31721 INVITE
Content-Length: 0


--end msg--
  17:20:39.105   pjsua_core.c  RX 793 bytes Response msg 183/INVITE/ 
cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP  
91.78.58.52 
:5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb
From: sip:johnsmith at sip.pctel.ru;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3
To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9
Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e
CSeq: 31721 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:012345678901 at 217.73.112.9:5060>
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 30663 30663 IN IP4 217.73.112.9
s=session
c=IN IP4 217.73.112.9
t=0 0
m=audio 18218 RTP/AVP 117 8 0 3 101
a=rtpmap:117 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--end msg--
  17:20:39.105    pjsua_app.c  Call 0 state changed to EARLY (183  
Session Progress)
  17:20:39.128   strm0x865b74  VAD temporarily disabled
  17:20:39.129   strm0x865b74  Encoder stream started
  17:20:39.129   strm0x865b74  Decoder stream started
  17:20:39.129  pjsua_media.c  Media updates, stream #0: iLBC (sendrecv)
  17:20:39.129   conference.c  Port 3 (sip:012345678901 at sip.pctel.ru)  
transmitting to port 0 (Built-in Microphone)
  17:20:39.129   conference.c  Port 0 (Built-in Microphone)  
transmitting to port 3 (sip:012345678901 at sip.pctel.ru)
  17:20:39.129    pjsua_app.c  Media for call 0 is active
  17:20:39.131   Master/sound  Underflow, buf_cnt=0, will generate 1  
frame
  17:20:39.778   strm0x865b74  VAD re-enabled
  17:20:49.469   pjsua_core.c  RX 820 bytes Response msg 200/INVITE/ 
cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP  
91.78.58.52 
:5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb
Record-Route: <sip:217.73.112.14;lr=on>
From: sip:johnsmith at sip.pctel.ru;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3
To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9
Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e
CSeq: 31721 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:012345678901 at 217.73.112.9:5060>
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 30663 30664 IN IP4 217.73.112.9
s=session
c=IN IP4 217.73.112.9
t=0 0
m=audio 18218 RTP/AVP 8 117 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:117 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--end msg--
  17:20:49.470    pjsua_app.c  Call 0 state changed to CONNECTING
  17:20:49.470    inv0x85bc64  SDP negotiation done, message body is  
ignored
  17:20:49.470   pjsua_core.c  TX 388 bytes Request msg ACK/cseq=31721  
(tdta0x86c200) to UDP 217.73.112.14:5060:
ACK sip:012345678901 at 217.73.112.9:5060 SIP/2.0
Via: SIP/2.0/UDP  
91.78.58.52:5060;rport;branch=z9hG4bKPjnUzGc4KND2SpWxshSKHX8DMbQ1numik6
Max-Forwards: 70
From: sip:johnsmith at sip.pctel.ru;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3
To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9
Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e
CSeq: 31721 ACK
Route: <sip:217.73.112.14;lr>
Content-Length:  0


--end msg--
  17:20:49.470    pjsua_app.c  Call 0 state changed to CONFIRMED
  17:20:49.616   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.626   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.641   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.668   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.684   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.704   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.724   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.741   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.761   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.784   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.804   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.824   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.842   strm0x865b74  Bad RTP pt 8 (expecting 117)
  17:20:49.861   strm0x865b74  Bad RTP pt 8 (expecting 117)




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